Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
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: 212-812-1203
https://www.aceinnovative.com
On 7/5/23 11:58, Michael Ulitskiy wrote:
Hello,
Anyone? I have hard time to believe this is not possible with chan_pjsip.
Anyway, may I ask how people handle the following scenario which I
imagine should be quite common:
- I have internal extensions talk to each other using g722. so their
codec setting (with chan_sip now) is "allow=g722,ulaw"
- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence
codec selection on calling channel and the calls set 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
originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only
works on outbound (called) channel and has no affect on calling
channel. My experiments and function documentation (which says “Media
and codec offerings to be set on an outbound SIP channel prior to
dialing.”) seem to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s
equivalent of ${SIP_CODEC_INBOUND}? Or, in other words, what are we
supposed to do to influence /calling/ channel codec selection from
dialplan?
I’m working with asterisk 20.3.0.
Thank you,
Michael
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Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
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, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.aceinnovative.com On 7/6/23 12:47, Joshua C. Colp wrote: On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy wrote: Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. That functionality is not implemented as of this time. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com <http://www.sangoma.com> and www.asterisk.org <http://www.asterisk.org> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
After I have re-read the "PJSIP Advanced Codec negotiation" document, it
occurred to me that the desired behavior should actually happen
automatically, just due to the codec negotiation logic, but it looks
like asterisk doesn't actually follow the described logic which is
likely a bug.
Can you please follow with me through a simple sip call and see if I'm
missing something or asterisk actually doesn't do what it's supposed to do?
Here's the codec negotiation config:
CLI> pjsip show endpoint A
...
codec_prefs_incoming_answer : prefer:pending,
operation:intersect, keep:all, transcode:allow
codec_prefs_incoming_offer : prefer:pending,
operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_answer : prefer:pending,
operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_offer : prefer:pending, operation:union,
keep:all, transcode:allow
All endpoints have the same default config above.
Let's go over simplest scenario: A calls B.
A is configured with g722 and ulaw (allow=!all,g722,ulaw) and B is
configured with ulaw and alaw (allow=!all,ulaw,alaw)
1. codec_prefs_incoming_offer: A sends INVITE to asterisk with codecs in
SDP g722,g729,g711u,g711a:
...
m=audio 2266 RTP/AVP 9 18 0 8 101.
a=rtpmap:9 G722/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8
- according to Advanced Codec Negotiation logic now we have:
- pending=g722,g729,ulaw,alaw
- configured=g722,ulaw
Applying operation "intersect" the resulting resolved topology is
"g722,ulaw" which is sent to the core
2. codec_prefs_outgoing_offer: Outgoing channel driver receives the
offer from the core
- pending=g722,ulaw
- configured=ulaw,alaw
Applying operation "union" the resulting resolved topology should
be "g722,ulaw,alaw" which should be sent
to B in the outgoing INVITE. What I see is actually sent in
outgoing INVITE is "ulaw,alaw":
...
m=audio 41506 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
...
So this is the 1st point where codec negotiation doesn't work as
expected
3. codec_prefs_incoming_answer: B replies with "200 OK" which contains
only ulaw codec:
...
m=audio 2226 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
...
- pending: ulaw
- configured: ulaw,alaw (it's result of step 2. it should be
g722,ulaw,alaw but actually is ulaw,alaw as described in step 2)
Applying operation "intersect" the resulting resolved topology is
"ulaw" which is sent to the core
4. codec_prefs_outgoing_answer: asterisk replies "200 OK" back to A
- pending: ulaw (from step 3)
- configured: g722,ulaw (from step 1)
Applying operation "intersect" the resulting resolved topology
should be "ulaw". What I see is actually sent in
"200 OK" is "g722,ulaw":
...
m=audio 43004 RTP/AVP 9 0 101.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
...
If I understand it correctly the result of codec negotiation 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.aceinnovative.com
On 7/5/23 11:58, Michael Ulitskiy wrote:
Hello,
Anyone? I have hard time to believe this is not possible with chan_pjsip.
Anyway, may I ask how people handle the following scenario which I
imagine should be quite common:
- I have internal extensions talk to each other using g722. so their
codec setting (with chan_sip now) is "allow=g722,ulaw"
- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence
codec selection on calling channel and the calls set 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
originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only
works on outbound (called) channel and has no affect on calling
channel. My experiments and function documentation (which says “Media
and codec offerings to be set on an outbound SIP channel prior to
dialing.”) seem to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s
equivalent of ${SIP_CODEC_INBOUND}? Or, in other
Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github pointer. 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 inability to avoid transcoding on outbound calls. To be more specific the scenario is as follows: 1. a phone initiates a call offering g722,g711 to asterisk 2. asterisk creates outbound call to carrier offering g711 only (carrier only supports g711) 3. carrier accepts the call and outbound call leg is now running on g711 4. asterisk accepts a phone's call with g722 since it's allowed on phone's endpoint and was indicated as preferred in phone's INVITE and now initial call leg is running on g722, resulting in transcoding This is very disappointing. Since developers announced their plans to drop chan_sip from future asterisk versions It's already been removed and won't be in any future major releases. If you still need chan_sip after removal, you can continue adding it from out of tree and building it. I maintain a working version of it out of tree. I was under impression that chan_pjsip has reached feature paritiy with chan_sip. It has mostly, but not completely, no. What is needed is an ability to tell asterisk which codecs are allowed to be included in "200 OK" asterisk sends back to the phone. I guess we need to submit a feature request. How do we go about it these days? I'm not sure about the particulars of this issue at all, but to answer the question at hand, there's a repo for it: https://github.com/asterisk/asterisk-feature-requests.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
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 inability to avoid transcoding on outbound calls. To be more specific the scenario is as follows: 1. a phone initiates a call offering g722,g711 to asterisk 2. asterisk creates outbound call to carrier offering g711 only (carrier only supports g711) 3. carrier accepts the call and outbound call leg is now running on g711 4. asterisk accepts a phone's call with g722 since it's allowed on phone's endpoint and was indicated as preferred in phone's INVITE and now initial call leg is running on g722, resulting in transcoding This is very disappointing. Since developers announced their plans to drop chan_sip from future asterisk versions I was under impression that chan_pjsip has reached feature paritiy with chan_sip. What is needed is an ability to tell asterisk which codecs are allowed to be included in "200 OK" asterisk sends back to the phone. I guess we need to submit a feature request. How do we go about it these days? Thanks, Michael On 7/5/23 14:59, Michael Maier wrote: Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know, there is no working solution as of now. I discussed this problem years ago already here but unfortunately nothing usable happened so far (which I would know off). The 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_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between internal extensions naturally happen over g722 as its their preferred codec - for external calls I now set SIP_CODEC_INBOUND=ulaw to influence codec selection on calling channel and the calls set up using ulaw end-to-end Can somebody please advise how to achieve the same with chan_pjsip? Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
Anyone? I have hard time to believe this is not possible with chan_pjsip.
Anyway, may I ask how people handle the following scenario which I
imagine should be quite common:
- I have internal extensions talk to each other using g722. so their
codec setting (with chan_sip now) is "allow=g722,ulaw"
- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence codec
selection on calling channel and the calls set 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
originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only
works on outbound (called) channel and has no affect on calling
channel. My experiments and function documentation (which says “Media
and codec offerings to be set on an outbound SIP channel prior to
dialing.”) seem to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s
equivalent of ${SIP_CODEC_INBOUND}? Or, in other words, what are we
supposed to do to influence /calling/ channel codec selection from
dialplan?
I’m working with asterisk 20.3.0.
Thank you,
Michael
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New to Asterisk? Start here:
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[asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
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 and function documentation (which says “Media and codec
offerings to be set on an outbound SIP channel prior to dialing.”) seem
to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s equivalent
of ${SIP_CODEC_INBOUND}? Or, in other words, what are we supposed to do
to influence /calling/ channel codec selection from dialplan?
I’m working with asterisk 20.3.0.
Thank you,
Michael
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New to Asterisk? Start here:
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[asterisk-users] asterisk 11.20.0 segfaults, but no core dump produced
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 to avoid it? Thanks, Michael-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk JIRA notifications
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 of years, as I was busy with other projects, and something has changed in that time. Now I'm not receiving any email notification and I can't find any settings to change it. I have my email address set in my profile and in preferences I have setting named "My Changes" set to "Notify me". Can't see anything else that may be related. Is there anything else I'm missing? Is it working at all? Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP realtime: lots of problems
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_endpoints_v is postgresql view. 1. The biggest problem: if I have small number of endpoints (roughly up to a 100) then asterisk loads ok and pjsip seems to be working ok (with other problems described below). If I have larger number of endpoints (several hundred) then intermittently (but often) asterisk just hangs during loading. Attempting to start asterisk with console (-c) it never reaches the user prompt. pjsip isn't functional (doesn't reply to any sip messages). the only way out is to kill asterisk. It looks like I'm hitting some limit here. Am I doing something wrong? Is there any config option I'm missing? 2. When it loads ok then it performs initial load of all endpoints individually. Looking at postgres log it does the following: SELECT * FROM pjsip_endpoints_v WHERE id LIKE '%' ORDER BY id SELECT * FROM pjsip_endpoints_v WHERE id = 'ep1' SELECT * FROM pjsip_endpoints_v WHERE id = 'ep2' ... SELECT * FROM pjsip_endpoints_v WHERE id = 'epN' Needless to say it's extremely inefficient with larger number of endpoints. After this initial load it seems to work correctly - loading endpoints on demand and caching them if I configure caching. Is there a way to disable this initial load? I want it to load endpoints on demand only. 3. When pjsip receives sip message and tries to match it to endpoint by 'From' username it initially performs lookup for 'username@domain' and if it fails it falls back to lookup by username only. Looking at the postgres log it looks like the following: SELECT * FROM pjsip_endpoints_v WHERE id = 'ep1@domain' SELECT * FROM pjsip_endpoints_v WHERE id = 'ep1' In my environment the domain part may be different (depending on which proxy the user is terminated) and I want to perform lookup on userpart only. Is there a way to tell pjsip to ignore the domain? Otherwise it doubles the number of queries per request and that's again extremely inefficient. I'd appreciate any help and/or pointers. Thanks a lot, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP realtime: lots of problems
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 change anything if you use odbc instead of pgsql? > I did some testing with chan_sip/pgsql and had much less problems > when pgsql was replaced by odbc. for some reason i can't make it work. i've installed and configured odbc. asterisk tells me it's connected: *CLI> odbc show ODBC DSN Settings - Name: asterisk DSN:asterisk Last connection attempt: 1969-12-31 19:00:00 Pooled: No Connected: Yes i've changed extconfig.conf as follows: ps_endpoints => odbc,asterisk Now asterisk logs the following: WARNING[966]: config.c:2905 find_engine: Realtime mapping for 'ps_endpoints' found to engine 'odbc', but the engine is not available and don't see any queries in database log. Am I doing something wrong? > > > 1. The biggest problem: if I have small number of endpoints (roughly up to > > a 100) then > > asterisk loads ok and pjsip seems to be working ok (with other problems > > described below). > > If I have larger number of endpoints (several hundred) then intermittently > > (but often) asterisk > > just hangs during loading. Attempting to start asterisk with console (-c) > > it never reaches the user > > prompt. pjsip isn't functional (doesn't reply to any sip messages). > > Did you try "core show locks" in this situation? Yes, it seems to deadlock in res_config_pgsql. I've opened an asterisk issue here: https://issues.asterisk.org/jira/browse/ASTERISK-25455 Will see where it goes. Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom phone registering
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 remote. > Other devices are registering just fine, just not my polycom phones. > > Today I notices that ONE registered, but it grabbed port 5060. > > 1004/1004 12.215.64.135D Yes > Yes55068 > 1006/1006 12.215.64.135D Yes > Yes55066 > 401/401 (Unspecified)D Yes > Yes0 > 510/510 (Unspecified)D Yes > Yes0 > 511/511 12.215.64.135D Yes > Yes5060 > 524/524 (Unspecified)D Yes > Yes0 > 535/535 (Unspecified)D Yes > Yes0 > 537/537 (Unspecified)D Yes > Yes0 > > The 1XXX are non polycom phones and are working just fine. The other devices > are polycom phones and only one is registering. > > Each has this exact definition but unique to extension of course: > > [524] > type=friend > defaultname=524 > defaultuser=524 > secret= > dtmfmode=RFC2833 > host=dynamic > description=Polycom 0004f2323292 > > context=smvoice-sip > rtptimeout=60 > rtpholdtimeout=60 > rtpkeepalive=60 > callerid="Jerry" > qualify=no > canreinvite=yes > timezone=1 > nat=force_rport,comedia > disallow=all > allow=g722 > allow=ulaw > allow=alaw > > I am getting "401 unathorized" as the response going to the polycom phones > - then nothing back from the phone until the next registration attempt. > > How do I either make the polycom ONLY use port 5060 or tell teh polycom its > OK to use another port? > > I tried takign off the force_rport and just leave comedia but that did not > make a > difference either. > > thanks, > > Jerry-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP: how to retrieve underlying SIP Call-ID
Hello Matt,
That works (CHANNEL(pjsip,call-id)). Thanks.
I'm still interested to know if PJSIP_HEADER is supposed to be able to read
headers of the outbound channel.
I'm also very interested to know if there's any way in asterisk to access
headers received in "200 OK" 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 <mulits...@acedsl.com> wrote:
> > 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
> > PJSIP_HEADER
> >
> > seem to be unable to read headers for outbound channel.
> >
> >
> >
> > Here's what I do:
> >
> >
> >
> > [xyz]
> >
> > exten => 999,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
> >
> > same =>
> > n,Dial(PJSIP/xyz011101/sip:xyz011101@:5060,30,b(_pre_dial,s,1))
> >
> > exten => h,1,NoOp()
> >
> >
> >
> > [_pre_dial]
> >
> > exten => s,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
> >
> > same => n,Set(CHANNEL(hangup_handler_push)=_hangup,s,1())
> >
> > same => n,Return
> >
> >
> >
> > [_hangup]
> >
> > exten => s,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
> >
> > same => n,Return
> >
> >
> >
> >
> >
> > Here's the result:
> >
> > -- Executing [999@xyz:1] NoOp("PJSIP/poly_650_2_01-006f", "Call-ID:
> > e3e249e5-7e8941dd-da386565@192.168.100.238") in new stack
> >
> > -- Executing [999@xyz:2] Dial("PJSIP/poly_650_2_01-006f",
> > "PJSIP/xyz011101/sip:xyz011101@:5060,30,b(_pre_dial,s,1)")
> > in new stack
> >
> > -- PJSIP/xyz011101-0070 Internal Gosub(_pre_dial,s,1) start
> >
> > -- Executing [s@_pre_dial:1] NoOp("PJSIP/xyz011101-0070", "Call-ID: ")
> > in new stack
> >
> > -- Executing [s@_pre_dial:2] Set("PJSIP/xyz011101-0070",
> > "CHANNEL(hangup_handler_push)=_hangup,s,1()") in new stack
> >
> > -- Executing [s@_pre_dial:3] Return("PJSIP/xyz011101-0070", "") in new
> > stack
> >
> > == Spawn extension (xyz, 999, 1) exited non-zero on
> > 'PJSIP/xyz011101-0070'
> >
> > -- PJSIP/xyz011101-0070 Internal Gosub(_pre_dial,s,1) complete
> > GOSUB_RETVAL=
> >
> > -- Called PJSIP/xyz011101/sip:xyz011101@:5060
> >
> > == Using SIP RTP Audio TOS bits 184
> >
> > -- PJSIP/xyz011101-0070 is ringing
> >
> > -- PJSIP/xyz011101-0070 Internal Gosub(_hangup,s,1) start
> >
> > -- Executing [s@_hangup:1] NoOp("PJSIP/xyz011101-0070", "Call-ID: ") in
> > new stack
> >
> > -- Executing [s@_hangup:2] Return("PJSIP/xyz011101-0070", "") in new
> > stack
> >
> > == Spawn extension (xyz, 999, 1) exited non-zero on
> > 'PJSIP/xyz011101-0070'
> >
> > -- PJSIP/xyz011101-0070 Internal Gosub(_hangup,s,1) complete
> > GOSUB_RETVAL=
> >
> > == Spawn extension (xyz, 999, 2) exited non-zero on
> > 'PJSIP/poly_650_2_01-006f'
> >
> > -- Executing [h@xyz:1] NoOp("PJSIP/poly_650_2_01-006f", "") in new stack
> >
> >
> >
> > As you can see I can get Call-ID of inbound channel, but I receive null for
> > the outbound channel in both pre-dial and hangup handlers.
> >
> >
> >
> > So my question is if there's a way to retrieve SIP Call-ID for outbound
> > channels?
> >
> > Also the 2nd question is if PJSIP_HEADER is supposed to be able to read
> > headers of the outbound channel?
> >
>
> Hi Michael -
>
> While you can use PJSIP_HEADER, the ability to retrieve the SIP
> Call-ID through the CHANNEL function on a PJSIP channel was actually
> just added in 13.6.0, and should be in the latest RC (13.6.0-rc2 [2]).
>
> In either case, you're using a function as opposed to some
> application, which means you do need to call the functions on the
> specific channel. To get access to the outbound channel, you can use a
> pre-dial handler's 'b' option [3]. The Call-ID *should* be set up on
> the underlying invite session in the PJSIP dialog, even though it
> hasn't been transmitted yet.
>
> Matt
>
> [1] https://gerrit.asterisk.org/#/c/1204/
> [2] http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0-rc2
> [3] https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
>
> --
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[asterisk-users] PJSIP: how to retrieve underlying SIP Call-ID
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 PJSIP_HEADER
seem to be unable to read headers for outbound channel.
Here's what I do:
[xyz]
exten => 999,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
same =>
n,Dial(PJSIP/xyz011101/sip:xyz011101@:5060,30,b(_pre_dial,s,1))
exten => h,1,NoOp()
[_pre_dial]
exten => s,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
same => n,Set(CHANNEL(hangup_handler_push)=_hangup,s,1())
same => n,Return
[_hangup]
exten => s,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
same => n,Return
Here's the result:
-- Executing [999@xyz:1] NoOp("PJSIP/poly_650_2_01-006f", "Call-ID:
e3e249e5-7e8941dd-da386565@192.168.100.238") in new stack
-- Executing [999@xyz:2] Dial("PJSIP/poly_650_2_01-006f",
"PJSIP/xyz011101/sip:xyz011101@:5060,30,b(_pre_dial,s,1)") in
new stack
-- PJSIP/xyz011101-0070 Internal Gosub(_pre_dial,s,1) start
-- Executing [s@_pre_dial:1] NoOp("PJSIP/xyz011101-0070", "Call-ID: ")
in new stack
-- Executing [s@_pre_dial:2] Set("PJSIP/xyz011101-0070",
"CHANNEL(hangup_handler_push)=_hangup,s,1()") in new stack
-- Executing [s@_pre_dial:3] Return("PJSIP/xyz011101-0070", "") in new
stack
== Spawn extension (xyz, 999, 1) exited non-zero on 'PJSIP/xyz011101-0070'
-- PJSIP/xyz011101-0070 Internal Gosub(_pre_dial,s,1) complete
GOSUB_RETVAL=
-- Called PJSIP/xyz011101/sip:xyz011101@:5060
== Using SIP RTP Audio TOS bits 184
-- PJSIP/xyz011101-0070 is ringing
-- PJSIP/xyz011101-0070 Internal Gosub(_hangup,s,1) start
-- Executing [s@_hangup:1] NoOp("PJSIP/xyz011101-0070", "Call-ID: ") in
new stack
-- Executing [s@_hangup:2] Return("PJSIP/xyz011101-0070", "") in new
stack
== Spawn extension (xyz, 999, 1) exited non-zero on 'PJSIP/xyz011101-0070'
-- PJSIP/xyz011101-0070 Internal Gosub(_hangup,s,1) complete
GOSUB_RETVAL=
== Spawn extension (xyz, 999, 2) exited non-zero on
'PJSIP/poly_650_2_01-006f'
-- Executing [h@xyz:1] NoOp("PJSIP/poly_650_2_01-006f", "") in new stack
As you can see I can get Call-ID of inbound channel, but I receive null for the
outbound channel in both pre-dial and hangup handlers.
So my question is if there's a way to retrieve SIP Call-ID for outbound
channels?
Also the 2nd question is if PJSIP_HEADER is supposed to be able to read headers
of the outbound channel?
Thanks,
Michael
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[Asterisk-Users] Determining that call was transferred
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 experiments seem to confirm it, but I'm not sure it's true in 100% of the cases. Thanks, 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
[Asterisk-Users] question about type=user in sip.conf
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-proxy-out] type=peer username=user secret=secret .. [sip-proxy] type=user context=from-proxy The question is how asterisk determines that the call is from sip-proxy? Whatever I do all incoming calls coming from sip-proxy (or from any other sip device not registered locally) get into sip context (default context) instead of sip-proxy context. Could anybody enlighten me on this or point out to some documentation? Thanks, 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
Re: [Asterisk-Users] Problems with native h323 channel on Asterisk RC2: No early audio and codec negotiation issues
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, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2 for example), in which audio would begin before the call was answered. Early audio is useful i order to provide the calling user with remote end ringback as well as recorded announcements, etc. You may want to look at/try this http://bugs.digium.com/bug_view_page.php?bug_id=562 2) The codec capabilities that Asterisk sends seem strange. No matter which codecs we set in the h323.conf file, G711 is the only codec that is sent in the capabilities. In order to use any other codec, we have to enable only the needed codec and disable all others. Again, this problem did not exist in older * versions, like 0.9.2 and it's limiting the capabilities of Asterisk in H323. Has anyone dealt with this problem successfully? Best regards, -- Vlasis Hatzistavrou. ___ 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] chan_h323 doesn't pass audio before call is answered
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 unanswered call. All they hear is just continuous ringtone as though remote phone is ringing but noone answering. I confirmed that gateway (Lucent MAX TNT) and GNUGK is not a problem because I can hear those announcements on another h323 endpoint registered directly with GNUGK. Also it did work with previous versions of asterisk (up to 0.9.1) with some 3d-party patch downloaded at some point from bugtracker. Now the patch cannot be correctly applied to asterisk-1.0-RC2 and my attempts to manually apply it failed. Therefore I would like to request some help on solving this issue. Does it work for anybody? May be with different PSTN gateway? Do you have any suggestions on fixing it? In addition as we are considering to get some Cisco PSTN/SIP gateway, I would like to ask if it's also the issue with SIP driver? Thanks a lot, 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
Re: [Asterisk-Users] Identifying which call an event belongs to
Unfortunately it doesn't help. If I specify actionid then I'll recevie Response: success or error with that actionid, but events will still flow without it. Any other ideas? Thanks, Michael On Wednesday 04 August 2004 07:15 pm, Nicolas Gudino wrote: Hello, On Wed, 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 how to associate a call and uniqueid field of event. Let's say I send the following manager command: action: originate channel: sip/[EMAIL PROTECTED] callerid: 1212555 MaxRetries: 1 WaitTime: 10 Application: AGI Data: callback.agi|212125551212555 Try inserting in your originate command: ActionID: SOME_RANDOM_ID Then I'm receiving the following events: Uniqueid: 1091642334.98 Event: Newchannel Callerid: State: Down Channel: SIP/pbx1-fc4f And you would probably receive: Uniqueid: 1091642334.98 Event: Newchannel Callerid: State: Down Channel: SIP/pbx1-fc4f ActionID: SOME_RANDOM_ID I did not try this, but I know that ActionID is implemented in some manager commands. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ 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] Identifying which call an event belongs to
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 command: action: originate channel: sip/[EMAIL PROTECTED] callerid: 1212555 MaxRetries: 1 WaitTime: 10 Application: AGI Data: callback.agi|212125551212555 Then I'm receiving the following events: Uniqueid: 1091642334.98 Event: Newchannel Callerid: State: Down Channel: SIP/pbx1-fc4f Uniqueid: 1091642334.98 Event: Newcallerid Callerid: 1212555 Channel: SIP/pbx1-fc4f Uniqueid: 1091642334.98 Event: Newchannel Callerid: 1212555 State: Up Channel: SIP/pbx1-fc4f ... etc. The question is how do I associate event with the originated call. The could be other concurrent calls that also generate events and I don't seem to be able to find a way to distinguish them. Is it possible? Thank you, 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
Re: [Asterisk-Users] PSTN intercepted announcement
As I understand you have PRI connected to asterisk directly. In my case there is a h323 gateway between them and the h323 driver must recognize the not in service signal and made asterisk aware of it so that asterisk could relay the conditions/recorded messages to SIP phones. From my 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 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 (number is not in service etc.) when I'm calling a disconnected number through asterisk. AFAIK, A PRI is normally expected to signal number not in service conditions out of band, so * should be signalling the out of service condition in a manner appropriate for the channel type (as a recording, or as the most accurate protocol specific out of service response code available, casung the out of service condition to be indicated by the target equipment) With my SNOM phone, the PRI signals not in service or no route or whatever, then * relays that in the form of a SIP error response, then the SNOM phone executes an internal recording and shows the error on the display as well. This results in more reliable call handling all the way through, and some bandwidth savings. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 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 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 (number is not in service etc.) when I'm calling a disconnected number through asterisk. The phone just keeps ringing. I know everything's fine with my PSTN connection and gateway, because I have other H.323 endpoints connecting directly to gateway without asterisk involved and it works for them. It seems that somehow both available h323 drivers for asterisk cannot handle those messages. I did some experimenting and found that H.323 FastFtart must be enabled in order for this to work (without faststart enabled it doesn't work for h.323 endpoints too). I tried to explicitly enable it on both h323 and oh323 drivers, but it didn't work with asterisk anyway. For the case of OH323, can you send me more details (conf file, log, tracefile) to check it? I'm not a telecom professional and I'm stucked here. So I thought I'd ask for help here :) Has anybody noticed this? Any ideas? Thanks. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Already on the phone?
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 callwaiting from within the dialplan. I need to avoid the callwaiting behavior in some cases and pass the call to another extension if called extension is already in use. Is this possible with asterisk? I've tried chanisavail application, but since callwaiting is enabled it always returns true. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium cards just for timing
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 problems should be gone 'cause those cards provide some reliable timing. So I have no choice and wish to buy a cheapest Digium card just for timing. I have no PSTN ports, it's pure voip environment here. So my question is whether any Digium card would be ok or I have to buy some specific card? I'm looking at X100P card as it is the cheapest one. Would it be enough? Thank you. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards just for timing
Martin, Thanks a lot. The problem was a turned on silence suppression on cisco ata 186. Now it seems to work perfectly. Thanks to everybody else too. Michael On Tuesday 14 October 2003 05:04 pm, Martin Pycko wrote: With the musiconhold and SIP-SIP call it turnes out that you need to disable 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 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 problems should be gone 'cause those cards provide some reliable timing. So I have no choice and wish to buy a cheapest Digium card just for timing. I have no PSTN ports, it's pure voip environment here. So my question is whether any Digium card would be ok or I have to buy some specific card? I'm looking at X100P card as it is the cheapest one. Would it be enough? Thank you. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 backup proxy registration
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 Release Notes for software 4.2 (http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/relnote/phnrn42s.htm#58498) says: The SIP phone can register with a backup proxy to support Survivable Remote Site Telephony (SRST). If the main proxy goes down, the backup proxy has the registration information required to route calls successfully. It does register the 1st line. It makes absolutely no sense to me to register just the 1st line and abondone the others. Michael On Thursday 04 September 2003 11:38 pm, Shawn L. Djernes wrote: From What I understand of this feature it is only to keep the phone working not to provide full services. I think they intended it to be something like a less powerful router or a box at a remote site. This way if the primary server was took out by a virus or hardware 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: [Asterisk-Users] 7960 backup proxy registration 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 primary proxy failure. I've set in SIPDefault.cnt: proxy1_address: 192.168.1.10 proxy1_port: 5060 proxy_backup: 192.168.1.12 proxy_backup_port: 5060 The problem is that 7960 registers all the configured lines with primary proxy, but the line 1 only with backup proxy. It's not about registration failure. The phone doesn't even try to register other 5 lines. As a result if the primary proxy fails incoming calls work for the line 1 only. Has anybody managed to register all the lines with backup proxy? I'm running software 4.4, the last version before digital signature was introduced. Should I upgrade? Or may be I'm missing something in configuration? Thanks a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 backup proxy registration
Well, on the other hand Release Notes for software 4.2 (http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/relnote/phnrn42s.htm#58498) says: The SIP phone can register with a backup proxy to support Survivable Remote Site Telephony (SRST). If the main proxy goes down, the backup proxy has the registration information required to route calls successfully. It does register the 1st line. It makes absolutely no sense to me to register just the 1st line and abondone the others. Michael On Thursday 04 September 2003 11:38 pm, Shawn L. Djernes wrote: From What I understand of this feature it is only to keep the phone working not to provide full services. I think they intended it to be something like a less powerful router or a box at a remote site. This way if the primary server was took out by a virus or hardware 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: [Asterisk-Users] 7960 backup proxy registration 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 primary proxy failure. I've set in SIPDefault.cnt: proxy1_address: 192.168.1.10 proxy1_port: 5060 proxy_backup: 192.168.1.12 proxy_backup_port: 5060 The problem is that 7960 registers all the configured lines with primary proxy, but the line 1 only with backup proxy. It's not about registration failure. The phone doesn't even try to register other 5 lines. As a result if the primary proxy fails incoming calls work for the line 1 only. Has anybody managed to register all the lines with backup proxy? I'm running software 4.4, the last version before digital signature was introduced. Should I upgrade? Or may be I'm missing something in configuration? Thanks a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 backup proxy registration
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 non-responsive. That usually means if the proxy responds with even an icmp port unreachable, it is still responding and the phone won't fail over to the backup. To validate, I'd suggest disconnecting the primary proxy to see if the phone then registers with other servers. No, it's not the case. The phone seems to work properly. It recognizes primary proxy failure and send INVITEs to the backup proxy. It does it for all lines. It just doesn't register lines 2-6 with backup proxy, so inbound calls for the numbers configured on those lines fail. I was hoping that somebody more experienced than myself could confirm or refute it. May be somebody tried it with the latest software releases? Also, the v4.4 release notes for Open Caveats says... CSCea15061: Outbound Proxy reREGISTER fails due to incorret logic. The Resolved Caveats tend to suggest that some related problems were fixed in v4.4, so this must have been an issue with previous releases. Well, I guess outbound proxy is a different story. Thanks anyway. Michael -Original Message- 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 primary proxy failure. I've set in SIPDefault.cnt: proxy1_address: 192.168.1.10 proxy1_port: 5060 proxy_backup: 192.168.1.12 proxy_backup_port: 5060 The problem is that 7960 registers all the configured lines with primary proxy, but the line 1 only with backup proxy. It's not about registration failure. The phone doesn't even try to register other 5 lines. As a result if the primary proxy fails incoming calls work for the line 1 only. Has anybody managed to register all the lines with backup proxy? I'm running software 4.4, the last version before digital signature was introduced. Should I upgrade? Or may be I'm missing something in configuration? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 call segmentation fault
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 - sip, oh323-isdn, oh323-capi asterisk crash with segmentation fault. Channel driver was compiled with pwlib 1.5.0 and openh323 1.12.0 libs. Does anybody know solution ? WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Executing Dial(H323:31119, SIP/92) in new stack -- Called 92 -- SIP/92-e46b is ringing -- SIP/92-e46b is ringing -- SIP/92-e46b is ringing -- SIP/92-e46b is ringing -- SIP/92-e46b answered H323:31119 PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. 0:58.180 H245:8128d60 RTP_UDP No mediaControlChannel specified PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. Segmentation fault regads Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 backup proxy registration
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 primary proxy failure. I've set in SIPDefault.cnt: proxy1_address: 192.168.1.10 proxy1_port: 5060 proxy_backup: 192.168.1.12 proxy_backup_port: 5060 The problem is that 7960 registers all the configured lines with primary proxy, but the line 1 only with backup proxy. It's not about registration failure. The phone doesn't even try to register other 5 lines. As a result if the primary proxy fails incoming calls work for the line 1 only. Has anybody managed to register all the lines with backup proxy? I'm running software 4.4, the last version before digital signature was introduced. Should I upgrade? Or may be I'm missing something in configuration? Thanks a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers
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 channel drivers. Nice to have something like this. I 'll add it to the TODO features of asterisk-oh323. 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 registration attempts in the background? Didn't test it with chan_oh323. There is no such problem with chan_oh323. Thank you. Michael Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Secondary gatekeeper support by asterisk h323 drivers
On Tuesday 26 August 2003 05:26 pm, Jeremy McNamara wrote: H.323 is no longer part of my own network, so I no longer have a built in method to burn lots of time beating up a dying protocol and unfortunately, I have to dedicate myself to projects that directly generate revenue. I'll see what I can do, but I can make no promises. I understand. I hope it doesn't mean that chan_h323 is now unsupported. Sorry, Jeremy McNamara P.S. I could not duplicate any crash using my own (proprietary) gatekeeper. Plus, I cannot do a thing without debug information. This is output of asterisk startup with chan_h323 when gatekeeper is unavailable or reject registration: [EMAIL PROTECTED]:/etc/asterisk# asterisk -cd DEBUG[1024]: File config.c, Line 744 (__ast_load): No file to parse: /usr/local/asterisk/etc/asterisk/asterisk.conf Asterisk CVS-08/05/03-16:15:37, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] ... [chan_h323.so] = (The NuFone Network's Open H.323 Channel Driver) == Creating H.323 Endpoint == Setting default context to h323 == Adding alias pbxb to endpoint == Adding Prefix 1212555 to endpoint == H.323 listener started *** Error registering with gatekeeper 192.168.0.7. ERROR[1024]: File chan_h323.c, Line 1673 (load_module): Gatekeeper registration failed. WARNING[1024]: File loader.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, 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 channel drivers. Nice to have something like this. I 'll add it to the TODO features of asterisk-oh323. 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 registration attempts in the background? Didn't test it with chan_oh323. There is no such problem with chan_oh323. Thank you. Michael Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musiconhold interrupted sound
Just updated asterisk from cvs. It's now CVS-08/01/03-18:27:08. Also I've removed ztdummy module. It seems to be better now. Not perfect - some sound glitch still happens, but much better. Also I've noticed that it somehow depends on the mp3 itself. Some songs are played out almost perfectly, with others the problem shows up again. Now I can see the following line in the debug log at the moment of sound interruption: Aug 5 12:55:57 DEBUG[262160]: File rtp.c, Line 917 (ast_rtp_raw_write): Difference is 2152, ms is 289 Thanks everybody. Michael On Tuesday 05 August 2003 07:40 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. The result is the same whether h.323 channel participates in the call or it's pure sip-to-sip call. Did you try it without the ztdummy and zaprtc? I posted to Mark a couple of patches fixing MOH and prompt playback on machines without zaptel hardware. They are in CVS since end of July. I have no problem with H.323, SIP and Quicknet devices. Michael. Just updated to latest CVS and tried moh without ztdummy... ...works like a charm :) That means the only thing I seem to need ztdummy for now is meetme (without it I get the This is not a valid conference number message). Exactly. I wonder, how hard would be to remove this dependency? Jamie Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3xx SIP messages
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, 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musiconhold interrupted sound
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 detail. The sound interruption happens only when 2 endpoints are actually connected and one of them put the other on hold. If I set a special 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 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 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing. Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). None of them helped. Asterisk version is CVS-07/30/03-18:53:16 If anybody has any idea on how to fix it or what can be done to further troubleshoot it, I'd appreciate hearing from you. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musiconhold interrupted sound
Michael, With all due respect to both of you, it's not related to h.323 driver. The result is the same whether h.323 channel participates in the call or it's pure sip-to-sip call. Michael On Monday 04 August 2003 01:59 pm, Michael Manousos wrote: Use asterisk-oh323. 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 detail. The sound interruption happens only when 2 endpoints are actually connected and one of them put the other on hold. If I set a special 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 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 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing. Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). None of them helped. Asterisk version is CVS-07/30/03-18:53:16 If anybody has any idea on how to fix it or what can be done to further troubleshoot it, I'd appreciate hearing from you. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musiconhold interrupted sound
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. Hi Michael, No - you're not alone :) Glad to hear. Then there's a hope it's gonna be fixed :) I've had similar problems with SIP since I started experimenting with * nearly a year ago. I still get it on my current cvs system every now and then although it's not nearly as bad as it used to be. Sometimes the moh will start off ok but then seem to lose sync and break up after a minute or so, other times (like now ;) ) it seems perfect apart from the odd little glitch. I just put it down to the fact that I have no real zap devices (chan_capi sip only) and so rely on ztdummy for timing. I was hoping that when I got around to putting an digium fxs card in, the problem would disappear completely! The strange thing is that the fact whether ztdummy is loaded or not does not really affect sound quality in my case. I guess asterisk might not using/seeing it. Does it requires some special configuration? I may be wrong, but I seem to remember that Michael Manousos' oh323 channel driver was _not_ affected by this problem (this was before Jeremy McNamara's h323 channel was usable), maybe because it doesn't use the same * libraries as other voip channels, however it's been a while since I played with h323 so I don't know if that's still the case. This sounds really strange to me as I think the problem is central to asterisk, not to particular driver and behavior is exactly the same over sip and chan_h323. Ok. I'll try chan_oh323 and see if it makes any difference. Jamie Neil Versado I.T. Services Ltd. Thanks Jamie. Michael One more detail. The sound interruption happens only when 2 endpoints are actually connected and one of them put the other on hold. If I set a special 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 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 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing. Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). None of them helped. Asterisk version is CVS-07/30/03-18:53:16 If anybody has any idea on how to fix it or what can be done to further troubleshoot it, I'd appreciate hearing from you. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!! Ringback oh323
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 list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer on ATA186
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 my ignorance... Make sure that you have set the DialPlan on the ATA 186 so as not to grab the # (ie look for any # character pairs and change the second character or remove it). Where to do that? In the extensions.conf file? Now I have used Flash key to put the other part on hold and then dial to the new extension and after this one answer, I close the phone. It works in that way only if the last party is anything else, but not another ATA186. Does it really work this way for you? I thought asterisk cannot bridge together 2 channels if originating party hangs up. I mean if I press flash button to put one party on hold, then dial another extension and then hang up the two other extensions do not get connected but both calls get dropped. Only blind transfer with # key works for me. If it really works for you, would you mind to show your configuration? Thanks. Michael Thanks for your support, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer on ATA186
I meant that attended transfer doesn't work (at least for me) when I'm trying to transfer call to a different device too. Let's say I dial from ata186 another h323 endpoint. Put it on hold. Then dial my cell phone (ata - asterisk chan_h323 - h323/pstn gateway). Then if I hang up on ata the call to my cell phone drops. What interesting is that asterisk redial my cell on itself in a second or so and then I get connected to h323 endpoint. If before hanging up I press flash on ata186 to have 3way conference call, it works fine - 3 phones get connected, but then if I hang up on ata then 2 other parties don't stay connected but both get dropped. In the latter case asterisk doesn't redial either phone. I think I've seen in the development maillist that asterisk doesn't support attended call transfer yet (at least on voip channels). It would be nice if someone of the gurus confirm (or better disprove ;-)) this. Michael On Monday 28 July 2003 01:00 pm, Dan wrote: Hi, It works, bot ONLY when I try to transfer the call to another type of phone, like X-Lite or Cisco 7960. If the destination is an ATA too, it does not work because hanging-up is considered as a closed call only after 1 second in ATA (if less than 1s, the it is a flash function), but the transfer function in Asterisk tries to recall the first extension in less than 1 second, so during this short period of time, ATA based phone is bussy and cannot accept calls, so the call is redirected to the voicemail. One way to make this attended transfer work with ATA too, is to enter a minimum delay of 1 second in th transer function, but I don't know how to do it. Look at the ATA186 specification for extended SIP functions, at the address: http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataadmn/ata88sip/supp.pdf or as HTML ast: http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html It is stated that the attended transfer is done like that: Step 1 Press the flash button on the telephone handset to put the existing party on hold and get a dial tone. Step 2 Dial the telephone number to which the existing party is being transferred. Step 3 When the callee answers 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 unattended one works perfect. - Original Message - From: Michael Ulitskiy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 28, 2003 7:41 PM Subject: Re: [Asterisk-Users] Call transfer on ATA186 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 my ignorance... Make sure that you have set the DialPlan on the ATA 186 so as not to grab the # (ie look for any # character pairs and change the second character or remove it). Where to do that? In the extensions.conf file? Now I have used Flash key to put the other part on hold and then dial to the new extension and after this one answer, I close the phone. It works in that way only if the last party is anything else, but not another ATA186. Does it really work this way for you? I thought asterisk cannot bridge together 2 channels if originating party hangs up. I mean if I press flash button to put one party on hold, then dial another extension and then hang up the two other extensions do not get connected but both calls get dropped. Only blind transfer with # key works for me. If it really works for you, would you mind to show your configuration? Thanks. Michael Thanks for your support, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No callerid on outgoing call over chan_h323
Works great for me. Thanks a lot. Michael On Wednesday 23 July 2003 01:40 pm, Jeremy McNamara wrote: I have committed some changes to the Asterisk CVS. Update and see if what I did actually fixes the problem. I forsee a couple major issues with the way I implemented it, but 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 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 number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other issues with that driver. So I switched to chan_h323 and everything seems to work fine except that I cannot set callerid on outgoing h323 calls even with SetCallerID application. Any advices? Thanks a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No callerid on outgoing call over chan_h323
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 No problem, always glad to help. Are you, by chance, running a Gatekeeper? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No callerid on outgoing call over chan_h323
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 number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other issues with that driver. So I switched to chan_h323 and everything seems to work fine except that I cannot set callerid on outgoing h323 calls even with SetCallerID application. Any advices? Thanks a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation fault with chan_oh323
That is another problem I hope the developers would pay attention to. ulaw codec segfaulting when it is used by h323 side of connection for both incoming and outgoing calls. At least with chan_oh323. If I set alaw codec for h323 it works fine regardless of codec on SIP side. Michael On Thursday 17 July 2003 03:36 am, Mark Thompson wrote: This also happened to me when I was using the same codec with both oh323 and SIP, if I forced it to alaw on oh323 and ulaw on SIP the connection worked. I also tried h323 instead of oh323 which works okay but you have to use earlier 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'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 and then silently crashes (it launched as asterisk -cd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=phone number. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. Thank a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault with chan_oh323
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 and then silently crashes (it launched as asterisk -cd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=phone number. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. Thank a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
