Hi,
I'm trying to interconnect sip and h323 endpoints using asterisk
and asterisk crashes with segmentation fault whenever h323
connection needs to be established. It registers with gatekeeper ok though.
Here are the symptoms.
If the call initiated by SIP device, asterisk replies to it "Trying" a
versions of pwlib and openh323.
> Mark
>
> -Original Message-
> From: Michael Ulitskiy [mailto:[EMAIL PROTECTED]
> Sent: 16 July 2003 23:44
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Segmentation fault with chan_oh323
>
>
> Hi,
>
> I'
Hi,
Has anybody managed to get callerid properly set on a call from
local to asterisk SIP endpoint through h323-pstn gateway to a
regular phone.
I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it.
When I place a call to pstn I'm not receiving 12125551234 as the clid,
but a numbe
t have
> no facilities to setup to test all the various possibilities.
>
>
> Jeremy McNamara
>
>
>
> Michael Ulitskiy wrote:
>
> >Hi,
> >
> >Has anybody managed to get callerid properly set on a call from
> >local to asterisk SIP endpoint
Yes, I am. Any way I can help?
You can reach me at mdu113 at acedsl dot com
Michael
On Wednesday 23 July 2003 03:07 pm, Jeremy McNamara wrote:
>
> Michael Ulitskiy wrote:
>
> >Works great for me.
> >Thanks a lot.
> >
> >Michael
> >
> >
>
&g
On Monday 28 July 2003 12:24 pm, Dan wrote:
> Hi Iain,
>
> > The basic call transfer functions, set with the T and t options to the
> dial
> > application and triggered by pressing a # work fine for me.
> I have T and t options in dial application, but how can '#' be used for
> transfer.
> Escuse
swers the phone, you may consult with the callee
> and then transfer the existing party by hanging up your telephone handset.
>
> It works for me on ATA if the final destination is not an ATA too.
>
> Best regards,
> Dan
> P.S. I'm interested in the attended transfer. The
Hi,
I don't seem to be able to get music on hold to play normally.
The sound gets often interrupted with a few seconds of silence
then starts playing again. I'm using mpg123-0.59r and tried
mp3 files with different sample rates with no luck. If that matters,
endpoints are SIP ata186, SIP Cisco 79
Specify option 'r' to dial application.
Michael
On Friday 01 August 2003 07:13 pm, [EMAIL PROTECTED] wrote:
> Hi
>
> What command i need to use to make a call with oh323 and hear the
> ringback sound
>
> Thanks
>
>
> ___
> Asterisk-Users mailing li
ecial extension with
SetMusicOnHold application it seems to play just fine.
Please help!
Thank you.
Michael
On Friday 01 August 2003 04:58 pm, Michael Ulitskiy wrote:
> Hi,
>
> I don't seem to be able to get music on hold to play normally.
> The sound gets often interrupted with a
. It works great!
>
> Michael.
>
>
> Michael Ulitskiy wrote:
> > Hi again,
> >
> > Am I really the only one who's having this problem?
> > Music on hold playing like this is very annoying and
> > practically unusable.
> > One more detai
On Monday 04 August 2003 02:56 pm, Jamie Neil wrote:
> Quoting Michael Ulitskiy [EMAIL PROTECTED]:
> > Hi again,
> >
> > Am I really the only one who's having this problem?
> > Music on hold playing like this is very annoying and
> > practically unusable.
:40 am, Michael Manousos wrote:
> Jamie Neil wrote:
> > Quoting Michael Manousos:
> >
> >>Michael Ulitskiy wrote:
> >>
> >>>Michael,
> >>>
> >>>With all due respect to both of you, it's not related to h.323 driver.
> >&
Hi,
Does anyone know if asterisk can handle 3xx SIP responces?
I'm trying make it work with redirect server and it looks like
asterisk isn't going to send another invite, but treats "302 Moved
Temporarily" message as "Everyone is busy".
Thanks.
Michael
___
Thanks Mark.
Any plans on implementing full redirect functionality?
Michael
On Thursday 07 August 2003 06:06 pm, Mark Spencer wrote:
> He should treat the first part as a local extension.
>
> amark
>
> On Thu, 7 Aug 2003, Michael Ulitskiy wrote:
>
> > Hi,
> >
&g
Hi,
I'm wondering if there are any plans on adding secondary gatekeeper
support to asterisk h323 channel drivers.
Also I've noticed that chan_h323 is crashing asterisk at startup if
primary gatekeeper is not available. Wouldn't it be a more correct
behavior if it doesn't crashing but continue re
Great! Thanks, Michael.
Jeremy, what do you think?
Michael
On Tuesday 26 August 2003 07:53 am, Michael Manousos wrote:
> Michael Ulitskiy wrote:
> > Hi,
> >
> > I'm wondering if there are any plans on adding secondary gatekeeper
> > support to asterisk h323 ch
c, Line 299 (ast_load_resource): chan_h323.so: load_module
failed, returning -1
== PWLib proces going down.
WARNING[1024]: File loader.c, Line 345 (load_modules): Loading module chan_h323.so
failed!
Segmentation fault
>
>
> Michael Ulitskiy wrote:
>
> >Great! Thanks, Micha
Hi,
I'm sorry to ask this question, but I thought I'd rather ask it here before
messing up with cisco.
Is anybody running cisco 7960 in redundant configuration?
I mean I want the phone to be registered with both primary and
backup proxy (asterisks) so that service continues to work in case of pri
re failure your office staff could
> still call for help.
>
> Shawn
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Michael Ulitskiy
> Sent: Thursday, September 04, 2003 18:10
> To: [EMAIL PROTECTED]
> Subject: [Aster
On Friday 05 September 2003 08:21 am, Rich Adamson wrote:
> I'm no where near an expert (or even very knowledgable on some of this stuff),
> but a fair number of machines (regardless of whether its a 7960 or whatever)
> will not fail over to secondary/backup gateways unless the primary is totally
>
If you are using ulaw codec, try change it to alaw.
oh323 currently has some problems with ulaw codec.
Michael
On Friday 05 September 2003 10:22 am, Marian Danisek wrote:
> hello,
> i have problem with oh323 channel driver (tryied differnet versions).
> when i try to make call between oh323 - s
If somebody's interested...
Cisco confirmed that current SIP images up to 5.3 cannot
register any lines other than line 1 with backup proxy.
I've submitted a feature request.
Michael.
On Friday 05 September 2003 10:35 am, Michael Ulitskiy wrote:
> Well, on the other hand Rele
Hello,
I finally got to look at chan_sip to chan_pjsip migration again. This
time I’m having problems with influencing codec selection on originating
(calling) channel. It looks like PJSIP_MEDIA_OFFER only works on
outbound (called) channel and has no affect on calling channel. My
experiments
t up using ulaw end-to-end
Can somebody please advise how to achieve the same with chan_pjsip?
Thanks,
Michael
On 6/30/23 09:30, Michael Ulitskiy wrote:
Hello,
I finally got to look at chan_sip to chan_pjsip migration again. This
time I’m having problems with influencing codec selection on
o
e priority is not high
enough. I need a solution, too. I understand that this behavior is a
nogo if you have a lot of calls because transcoding is expensive.
Thanks
Michael
On 05.07.23 at 17:58 Michael Ulitskiy wrote:
Hello,
Anyone? I have hard time to believe this is not possible with
chan_p
ointer.
Michael
On 7/5/23 16:46, aster...@phreaknet.org wrote:
On 7/5/2023 4:19 PM, Michael Ulitskiy wrote:
Hi Michael,
Thanks for the reply.
I was referring to the scenario you named as 'outbound broken'. I
didn't get to look at inbound call behavior yet, as I got stuck with
on in the
above scenario should be ulaw in both call legs, thus avoiding
transcoding, but actual asterisk behavior differs.
Am I missing something. What are your thoughts?
Thanks,
*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.
Oh, that's great. It wasn't clear from that page, at least not for me. :-(
Having it clearly stated on the document would save me (and probably
others) lots of time.
Thanks for clarifying it. Any idea on the timeframe of implementation?
*Michael Ulitskiy*
Ace Innovative Networks,
FYI: i've created a feature request to add SIP_CODEC_INBOUND equivalent
functionality to chan_pjsip:
https://github.com/asterisk/asterisk-feature-requests/issues/9
Let's see where it goes
*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-
Hello,
I've started to play with PJSIP and got stuck at the following problem.
I need to retrieve SIP Call-ID associated with PJSIP channel.
For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't
work for
outbound channel even in pre-dial or hangup handler. Whatever I do P
" reply,
as my proxy returns some important information there.
Thanks a lot,
Michael
On Tuesday, October 06, 2015 05:06:34 PM Matthew Jordan wrote:
> On Tue, Oct 6, 2015 at 3:25 PM, Michael Ulitskiy wrote:
> > Hello,
> >
> >
> >
> > I've started to play with
Hello,
I wonder if anybody is using PJSIP realtime in production environment?
I've started to play with it and encountered many problems.
Here's my config:
sorcery.conf:
[res_pjsip]
endpoint=realtime,ps_endpoints
extconfig.conf:
[settings]
ps_endpoints => pgsql,users,pjsip_endpoints_v
pjsip_end
On Thursday, October 08, 2015 08:02:19 PM Stefan Tichy wrote:
> Hello Michael
>
> On Thu, Oct 08, 2015 at 01:32:07PM -0400, Michael Ulitskiy wrote:
> >
> > extconfig.conf:
> > [settings]
> > ps_endpoints => pgsql,users,pjsip_endpoints_v
>
> Does it
It sounds like you have problems with your firewall. Your 401 replies don't
reach the phones.
On Thursday, October 08, 2015 02:50:24 PM Jerry Geis wrote:
> Do polycom phones not LIKE using something other than port 5060 ???
>
> I have five of them behind a firewall and my asterisk server is remo
Hello,
For some reason I'm not receiving any notifications from JIRA.
I've been using Mantis, then JIRA for a long time and I have always received
email notification when there was an activity in issues I've opened and/or
those I'm watching.
I haven't been using asterisk bugtracker for a couple
Hello,
I've had several occurences of asterisk segfault (exited on signal 11), but no
core dump produced.
asterisk workdir is /tmp, /tmp is world-writeable and asterisk was started as
"asterisk -f -I -vvv -g"
What else am I missing? Is there situations where core dump isn't produced? Any
way
Hi,
I've found that neither Michael Manousos patch nor ztdummy driver
do not fix musiconhold sound interruption problem up to acceptable quality
level. Sound is choppy here anyway.
It is my understanding (please correct me if I'm wrong) that if I have
a Digium card in my asterisk machine, these p
isable silence supporesion on your phones/gateways since the timing is
> taken from the coming stream (but only for musiconhold AFAIK)
>
> regards
> Martin
>
> On Tue, 14 Oct 2003, Michael Ulitskiy wrote:
>
> > Hi,
> >
> > I've found that neither Michae
Hi,
I'm wondering if there's a way within a dialplan or AGI to find out
if an extension (SIP client) is already in use and the
person is already on the phone?
By default the channel is assumed available and callwaiting tone
is transmitted to the called extension. AFAIK there's no way to turn
off
Paul Liew wrote:
> Michael,
>
> I've added a patch a week ago on to bugtracker to fix this - feel free to
> try it and let me know
>
> http://bugs.digium.com/bug_view_page.php?bug_id=408
>
> Paul
> - Original Message -
> From: "Michael Ulitsk
Paul,
I'm using Cisco 7960 phones. I did some more testing and it looks like
using chanisavail with SIP channel causes it loose inuse status.
I've removed chanisavail application from dialplan and now
I cannot reproduce the problem whether the call is on hold or not.
So you patch is probably fine.
Hi,
I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have
H.323 to PSTN
gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI.
Everything works fine with one exception. I seem to be unable to figure out why I
cannot hear
PSTN intercepted announcement
Michael,
I've sent all info off-list.
Thanks.
Michael
On Thursday 20 November 2003 09:53 am, Michael Manousos wrote:
> Michael Ulitskiy wrote:
> > Hi,
> >
> > I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I
> > have H.323 to
experience so far, oh323 driver does it, h323 does not.
Please correct me if I'm wrong.
Thanks
Michael
On Friday 21 November 2003 03:33 am, Josh Rollyson wrote:
> Michael Ulitskiy wrote:
>
> >Hi,
> >
> >I have asterisk functioning as SIP to H.323 gateway for local SIP
Hi,
I guess I need some help with management interface. I would like to watch
calls through the management interface, but I don't know how to identify
which call an event belongs to or in other words how to associate a call
and uniqueid field of event.
Let's say I send the following manager comma
2004-08-04 at 18:56, Michael Ulitskiy wrote:
> > Hi,
> >
> > I guess I need some help with management interface. I would like to watch
> > calls through the management interface, but I don't know how to identify
> > which call an event belongs to or in other words h
Hi,
I have the following topology:
PSTN/H323 gateway->GNUGK->chan_h323/chan_sip->SIP EP
Mostly everything works fine except chan_h323 is not passing
audio from PSTN before the call is answered and as a result users
can't hear PSTN announcements (like "the number is not in service")
that's played
On Thursday 16 September 2004 04:27 am, Vlasis Hatzistavrou wrote:
> Hello all,
>
> We have been testing Asterisk RC2 with the native H323 channel driver.
> We followed the instructions with the needed OpenH323 and PWLib versions
> and everything compiled ok. Operation of the driver seems ok, e
Hi,
I may be missing something here, but I don't really understand
how asterisk supposed to handle type=user.
Suppose I have the following config (mostly taken from default sip.conf.sample):
sip.conf:
context=sip ;default context for incoming calls
...
register => [EMAIL PROTECTED]
..
[sip-pro
Hi,
I wish to determine whether a call is original call or it was transferred by someone.
I need to do it within AGI script.
Can I consider the following statements true:
1. if agi_extension != agi_dnid then the call is a transferred call
2. the call was transferred by extension agi_extension
My
51 matches
Mail list logo
