Re: [asterisk-users] Some Good News for VoIP
On Tue, 16 Dec 2008 17:55:00 -0500, Drew Gibson d...@oanda.com wrote: http://www.theregister.co.uk/2008/12/16/infonetics_enterprise_telephony_numbers_q3_2008/ ... whilst... http://www.theregister.co.uk/2008/12/15/adsl_voip/ G ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libpri and NT-Point to multi-point
Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WTB: Digium 1 or 4 ports E1 Cards
Hi, I am looking to buy 2 used 1 or 4 ports E1 Cards. If you have one, would you please contact me? Thanks, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri and NT-Point to multi-point
If you are connecting to BRI lines then you should be TE - not NT. You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried the new release). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 08:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] libpri and NT-Point to multi-point Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri and NT-Point to multi-point
Hello Andrew, 2008/12/17 Andrew Thomas a...@datavox.co.uk If you are connecting to BRI lines then you should be TE - not NT. Yes of course, you're right. I was mostly referring to this : ISDN --BRI asterisk -BRI- legacy PBX Then, in this case, as legacy PBX has a set of TE-PtmP or TE-PtP interfaces, asterisk box should also include such NT-PtP or NT-PtmP interfaces. For instance, would you say that in the UK, most PBXes are using TE-PtP ? You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried the new release). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 08:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] libpri and NT-Point to multi-point Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri and NT-Point to multi-point
I have piggy backed a few PBX's off the back of a B410P (4 x BRI) card with no problems. The ones I used for testing were the Avaya IP Office, Siemens Hi-Path/Hi-Com and various old Panasonics. All I had to do was to turn on the 100ohm termination on my S0 ports (set as NT on the B410P of course). I actually have a similar set-up at the moment on our main asterisk system. 2 x BRI trunks (ports 1 2 set as TE ptp - for the DDI's etc) and 2 x BRI S0 bus (ports 3 4 set as NT ptmp) running and ISDN modem on each (good old Fritz! ones). So it is possible to run ptmp on NT ports using mISDN - just remember to turn on 100ohm termination on the ISDN card if you only have one device per port. HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 09:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] libpri and NT-Point to multi-point Hello Andrew, 2008/12/17 Andrew Thomas a...@datavox.co.uk If you are connecting to BRI lines then you should be TE - not NT. Yes of course, you're right. I was mostly referring to this : ISDN --BRI asterisk -BRI- legacy PBX Then, in this case, as legacy PBX has a set of TE-PtmP or TE-PtP interfaces, asterisk box should also include such NT-PtP or NT-PtmP interfaces. For instance, would you say that in the UK, most PBXes are using TE-PtP ? You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried the new release). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 08:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] libpri and NT-Point to multi-point Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RDNIS and asterisk
Where are you actually doing the diverting? In Asterisk at the telco
exchange?
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith
-- Sent: 17 December 2008 11:07
-- To: Asterisk Users Mailing List - Non-Commercial Discussion
-- Subject: Re: [asterisk-users] RDNIS and asterisk
--
-- No, ${exten} is the final destination number
--
-- myphone calls 123456, which is diverted to 22334455 would givc an
-- ${exten} of 22334455, but I wanted to know the 123456.
--
-- Julian
-- Andrew Thomas wrote:
-- Isn't that the ${exten} number? In other words, the number
called.
--
--
--
-- -- -Original Message-
-- -- From: asterisk-users-boun...@lists.digium.com
-- [mailto:asterisk-users-
-- -- boun...@lists.digium.com] On Behalf Of Tony Mountifield
-- -- Sent: 17 December 2008 10:17
-- -- To: asterisk-users@lists.digium.com
-- -- Subject: Re: [asterisk-users] RDNIS and asterisk
-- --
-- -- In article 49483005.8030...@dotr.com,
-- -- Julian Lyndon-Smith aster...@dotr.com wrote:
-- -- I have a couple of numbers that are diverted to a number
-- that is
-- -- conected to an isdn30 card, running asterisk 1.4.
-- --
-- -- eg.
-- --
-- -- 123456 = 22334455
-- -- 654321 = 22334455
-- --
-- -- What I would like to know is the number of the orginal
-- number
-- -- dialled
-- -- (123456 or 654321). I thought that RDNIS was the answer,
but
-- it
-- is
-- -- always coming up blank.
-- --
-- -- When I did a debug on the pri span, I saw the following
-- message
-- --
-- -- Unable to handle ROSE operation 15
-- --
-- -- is this the cause of my problem ?
-- --
-- -- Don't know about that error, but in the pri debug output,
did
-- you
-- see
-- -- any mention of the originally dialled number, or only the
-- translated
-- -- number?
-- --
-- -- If the originally dialled number is not presented in an
-- information
-- -- element somewhere, then it would be a bit of a challenge
for
-- Asterisk
-- -- to infer it! :-)
-- --
-- -- Just found this message, which seems to refer to the same
-- issue:
-- --
-- http://lists.digium.com/pipermail/asterisk-users/2007-
-- July/191858.html
-- --
-- -- If your original number does appear in a ROSE IE simlar to
-- that
-- shown
-- -- in
-- -- the above message, then it may be that libpri needs
updating
-- to
-- handle
-- -- it.
-- --
-- -- Can you get a second destination number on the same ISDN30
and
-- then
-- -- divert one of the original numbers to that instead?
-- --
-- -- Cheers
-- -- Tony
-- -- --
-- -- Tony Mountifield
-- -- Work: t...@softins.co.uk - http://www.softins.co.uk
-- -- Play: t...@mountifield.org - http://tony.mountifield.org
-- --
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[asterisk-users] PostgreSQL CDRs 1.6 (was: Re: 1.6 upgrade issues)
Chris Bagnall schrieb: Also seem to be getting some errors writing CDRs to a postgresql database. What errors precisely? How can you tell? I'm using the schema for pgsql from voip-info.org, which, again, has worked fine logging 1.2 and 1.4. Have there been any schema changes in 1.6 one needs to be aware of? Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to tell when a issue actually gets in a released version
Jerry Geis schrieb: This bug report http://bugs.digium.com/print_bug_page.php?bug_id=12038 apparently has been fixed. I dont see anything on the page saying what released version of asterisk this is in. How can I tell that? It (svnbot) says: U branches/1.4/main/dial.c r104841 | mmichelson | 2008-02-27 15:45:47 -0600 (Wed, 27 Feb 2008) | 17 lines Which means it has been commited to the 1.4 branch at revision 104841 on 2008-02-27 15:45:47 -0600. Any tag (aka. release) made of 1.4 after this date will have it. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RDNIS and asterisk
No, ${exten} is the final destination number
myphone calls 123456, which is diverted to 22334455 would givc an
${exten} of 22334455, but I wanted to know the 123456.
Julian
Andrew Thomas wrote:
Isn't that the ${exten} number? In other words, the number called.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Tony Mountifield
-- Sent: 17 December 2008 10:17
-- To: asterisk-users@lists.digium.com
-- Subject: Re: [asterisk-users] RDNIS and asterisk
--
-- In article 49483005.8030...@dotr.com,
-- Julian Lyndon-Smith aster...@dotr.com wrote:
-- I have a couple of numbers that are diverted to a number that is
-- conected to an isdn30 card, running asterisk 1.4.
--
-- eg.
--
-- 123456 = 22334455
-- 654321 = 22334455
--
-- What I would like to know is the number of the orginal number
-- dialled
-- (123456 or 654321). I thought that RDNIS was the answer, but it
is
-- always coming up blank.
--
-- When I did a debug on the pri span, I saw the following message
--
-- Unable to handle ROSE operation 15
--
-- is this the cause of my problem ?
--
-- Don't know about that error, but in the pri debug output, did you
see
-- any mention of the originally dialled number, or only the
translated
-- number?
--
-- If the originally dialled number is not presented in an
information
-- element somewhere, then it would be a bit of a challenge for
Asterisk
-- to infer it! :-)
--
-- Just found this message, which seems to refer to the same issue:
--
http://lists.digium.com/pipermail/asterisk-users/2007-July/191858.html
--
-- If your original number does appear in a ROSE IE simlar to that
shown
-- in
-- the above message, then it may be that libpri needs updating to
handle
-- it.
--
-- Can you get a second destination number on the same ISDN30 and
then
-- divert one of the original numbers to that instead?
--
-- Cheers
-- Tony
-- --
-- Tony Mountifield
-- Work: t...@softins.co.uk - http://www.softins.co.uk
-- Play: t...@mountifield.org - http://tony.mountifield.org
--
-- ___
-- -- Bandwidth and Colocation Provided by http://www.api-digital.com
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[asterisk-users] ael queue gosub already has PBX structure??
Hello,
I want that after client and queue member call would be established, cmd
queue runs some 'procedures' . So I am using cmd Queue option 'gosub'. This
is my example of ael :
context QUEUE {
_X. = {
Ringing();
Wait(4);
Answer();
Queue(${Queue},wr,,,60,,,check-record);
Hangup();
};
};
macro check-record() {
Set(MEMBERNUMBER=${CUT(MEMBERINTERFACE,@,1)});
Set(MEMBERNUMBER=${CUT(MEMBERNUMBER,/,2)});
return;
};
Everything works normal, but when the client's and queue call establishes ,
I get this WARNING:
-- Local/1...@cc-out-da9a;1 answered SIP/xxx.xxx.xx-12d132d0
[Dec 17 20:52:12] WARNING[3849]: pbx.c:3656 __ast_pbx_run:
SIP/sip.call.lt-12d132d0 already has PBX structure??
== Starting SIP/sip.call.lt-12d132d0 at check-record,s,0 failed so falling
back to exten 's'
-- Executing [...@check-record:1] Set(SIP/sip.call.lt-12d132d0,
MEMBERNUMBER=Local/123) in new stack
-- Executing [...@check-record:2] Set(SIP/sip.call.lt-12d132d0,
MEMBERNUMBER=123) in new stack
What I'm missing? Something wrong with ael syntax/structure ?
Thanks in advance
--
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Alcatel OXE + Asterisk as external IVR
2008/12/17 Artifex Maximus artife...@gmail.com On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote: 2008/12/17 Artifex Maximus artife...@gmail.com Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI I've talked with support person at Alcatel and he said that Q.931 cannot handle this situation because after calls leave OXE it does not know anything so I cannot hangup in Asterisk and call will use two channel. Is it right? He said that ABCF2 or Q.SIG is able handling this situation because Q.SIG is an extension to Q.931. I take some search on topic and find out that Asterisk's Q.SIG not fully implemented. Is Asterisk implementation enough for this kind of setup? What is needed is that the Asterisk box should either : - forward incoming call to the right endpoint, using a single channel, - open a second channel and remain in media path till it ends. Thanks for your answer! You are right and first option what I am looking for. I have asked support staff and sending back DTMF on open channel does not help. True ! I'm not an authority on this topic, but I would say that, as OXE and asterisk are connected through an E1/T1 link, - you must upgrade OXE and Asterisk to Q.SIG to get forwarding option (and check asterisk's QSIG supports Call Deflection), - casual PRI is enough if you stick with 2 channels option. Unfortunately I am not expert on this topic as well but second option is not good for us. The question is how good Asterisk's Q.SIG implementation for this task. That's the question ! Maybe someone else could help on this as I don't have much experience to share. If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. Often we have more than 15 calls at same time and that is why first option is not acceptable. you mean second option is not acceptable, don't you ? Bye, Zsolt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and NAT one way audio
Hello may situation is the next: Asterisk -- NAT1 (router)--- internet -- NAT2 (router) -- x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone to the x-lite. From the x-lite to de ip phone there is no audio. I have made port fordwarding in the router 1. I have opened ports 5060-5070 to SIP and 1-2 to RTP. In the sip.conf file I have externip, localnet, bindport, bindaddres, nat = yes etc. I have used wireshark to see the frames and there is rtp traffic in both direccion so i don´t know what the problem is. I have used the x-lite in the same LAN than the ip-phone and all is good so the two phones works fine. What is the problem? Thank you in advance -- Silvia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interesting problem
So what is the middle name that causes problems? atre you sure you don't
have strange characters in it, like spaces, nonprintables, weird
encodings, etc?
l.
In data Wed, 17 Dec 2008 00:35:04 +0100, Eve Ellen Cole
ec...@mail.plymouth.edu ha scritto:
I’ve got an interesting problem and am wondering if anyone can shed
light …
I am running Asterisk on RHEL Server release 5.2 connecting to an Avaya
Definity G3R via a Digium TE220.
Asterisk 1.4.20
Zaptel 1.4.4
Libpri 1.4.4
MySQL 5.0.45
Festival Speech Synthesis System: 1.95
We have about 4200 accounts in a MySQL db. Asterisk retrieves the user
information from the database, festival tts says the name, then the
caller can leave a voicemail, which is then emailed to the user. At
this time, the system only takes calls, no calls go out.
The problem is that at times Asterisk doesn’t release the channel.
Messages in the log file show indicate the channel is busy. The only
way I can find to get the channel to release is to restart Asterisk.
[Dec 16 14:39:06] DEBUG[11141] chan_zap.c: Ring requested on channel 0/1
already in use or previously requested on span 1. Attempting to
renegotiating channel.
Since this is happening on a regular basis, I’ve been doing some
troubleshooting and can now predictably cause this problem. It mainly
seems to happen with one particular mailbox, and festival seems to be a
factor. When this particular mailbox is dialed, Asterisk goes through
the dialplan up to and including the Festival(${FULLNAME}) step, but not
beyond. Just for yucks, I changed the fullname of the person with that
mailbox by taking out the middle name. All seems to work fine without
the middle name. If I put a middle initial or middle name, the channel
locks up again. I’ve wondered if Festival has a problem with the length
of the name, but there are other students with longer names and this
problem doesn’t occur with their extensions. Any thoughts?
Dialplan
exten = _5[14-9]XXX,1,Answer()
exten = _5[14-9]XXX,n,Playtones(ring)
exten = _5[14-9]XXX,n,MYSQL(Connect CONNID localhost asterisk
HG06e6kghpUjtGvnX asterisk)
exten = _5[14-9]XXX,n,MYSQL(Query RESULTID ${CONNID} Select 'fullname'
from voicemail_users Where mailbox=${EXTEN})
exten = _5[14-9]XXX,n,MYSQL(Fetch FETCHID ${RESULTID} FULLNAME)
exten = _5[14-9]XXX,n,MYSQL(Disconnect ${CONNID})
exten = _5[14-9]XXX,n,GotoIf($[${FETCHID} = 1]?connect:disconn)
exten = _5[14-9]XXX,n(connect),StopPlaytones()
exten = _5[14-9]XXX,n,Wait(2)
exten = _5[14-9]XXX,n,Playback(you-have-dialed)
exten = _5[14-9]XXX,n,Playback(the-mailbox)
exten = _5[14-9]XXX,n,Playback(for)
exten = _5[14-9]XXX,n,Festival(${FULLNAME})
exten = _5[14-9]XXX,n,VoiceMail(${ext...@students)
exten = _5[14-9]XXX,n,Playback(goodbye)
exten = _5[14-9]XXX,n,Hangup() exten =
_5[14-9]XXX,n(disconn),Zapateller()
exten = _5[14-9]XXX,n,Playback(you-dialed-wrong-number)
exten = _5[14-9]XXX,n,Playback(check-number-dial-again)
exten = _5[14-9]XXX,n,Playtones(congestion)
exten = _5[14-9]XXX,n,Wait(3)
exten = _5[14-9]XXX,n,Hangup()
--
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Re: [asterisk-users] ael queue gosub already has PBX structure??
Giedrius Augys wrote:
Hello,
I want that after client and queue member call would be established,
cmd queue runs some 'procedures' . So I am using cmd Queue option
'gosub'. This is my example of ael :
context QUEUE {
_X. = {
Ringing();
Wait(4);
Answer();
Queue(${Queue},wr,,,60,,,check-record);
Hangup();
};
};
macro check-record() {
Set(MEMBERNUMBER=${CUT(MEMBERINTERFACE,@,1)});
Set(MEMBERNUMBER=${CUT(MEMBERNUMBER,/,2)});
return;
};
Everything works normal, but when the client's and queue call
establishes , I get this WARNING:
-- Local/1...@cc-out-da9a;1 answered SIP/xxx.xxx.xx-12d132d0
[Dec 17 20:52:12] WARNING[3849]: pbx.c:3656 __ast_pbx_run:
SIP/sip.call.lt-12d132d0 already has PBX structure??
== Starting SIP/sip.call.lt-12d132d0 at check-record,s,0 failed so
falling back to exten 's'
-- Executing [...@check-record:1] Set(SIP/sip.call.lt-12d132d0,
MEMBERNUMBER=Local/123) in new stack
-- Executing [...@check-record:2] Set(SIP/sip.call.lt-12d132d0,
MEMBERNUMBER=123) in new stack
What I'm missing? Something wrong with ael syntax/structure ?
Thanks in advance
--
Pagarbiai / Best Regards,
Giedrius Augys
This is a bug you are experiencing, which I fixed recently in a series of
commits. Assuming you are using a 1.6 tag, the next build should have this
problem fixed.
Mark Michelson
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Re: [asterisk-users] top posting again [was: Re: CDR Design]
OR Q: What is the most annoying thing in email? Q: What is the most annoying thing in email? A: Top-posting. Q: What is the most annoying thing in email? A: Top-posting. Q: Why is top-posting such a bad thing? Q: What is the most annoying thing in email? A: Top-posting. Q: Why is top-posting such a bad thing? A: Because it messes up the order in which people normally read text. In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gergo Csibra Sent: Friday, December 05, 2008 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] Friday, December 5, 2008, 2:49:59 PM, Andrew wrote: Address added to spam filter. Please do NOT e-mail me again. A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail? -- Best regards, Gergomailto:csi...@gmail.com OR Q: What is the most annoying thing in email? Q: What is the most annoying thing in email? A: Top-posting. Q: What is the most annoying thing in email? A: Top-posting. Q: Why is top-posting such a bad thing? Q: What is the most annoying thing in email? A: Top-posting. Q: Why is top-posting such a bad thing? A: Because it messes up the order in which people normally read text. In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RDNIS and asterisk
-- Where are you actually doing the diverting? In Asterisk at the telco -- exchange? ...or at... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
Bill Andersen wrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill Not when you take the time to properly trim your reply it's not. BK ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri and NT-Point to multi-point
I would say the 'norm' in the UK is TE-ptp and NT-ptp or NT-ptmp (depends what is on the end of the port(s)). If using NT-ptmp, then a 100ohm resistor is usually needed in the circuit somewhere - aka ISDN balun - (unless the card has this facility - like the B410P has). HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 10:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] libpri and NT-Point to multi-point 2008/12/17 Andrew Thomas a...@datavox.co.uk I have piggy backed a few PBX's off the back of a B410P (4 x BRI) card with no problems. The ones I used for testing were the Avaya IP Office, Siemens Hi-Path/Hi-Com and various old Panasonics. All I had to do was to turn on the 100ohm termination on my S0 ports (set as NT on the B410P of course). I actually have a similar set-up at the moment on our main asterisk system. 2 x BRI trunks (ports 1 2 set as TE ptp - for the DDI's etc) and 2 x BRI S0 bus (ports 3 4 set as NT ptmp) running and ISDN modem on each (good old Fritz! ones). Fine, so, using this setup as an example, would say the norm in the UK, is to connect to ISDN-BRI in ptp (reading from ports 1 and 2 configuration), or to connect using ptmp (reading from ports 3 and 4 configuration, dedicated to isdn modems) ? So it is possible to run ptmp on NT ports using mISDN - just remember to turn on 100ohm termination on the ISDN card if you only have one device per port. HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 09:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] libpri and NT-Point to multi-point Hello Andrew, 2008/12/17 Andrew Thomas a...@datavox.co.uk If you are connecting to BRI lines then you should be TE - not NT. Yes of course, you're right. I was mostly referring to this : ISDN --BRI asterisk -BRI- legacy PBX Then, in this case, as legacy PBX has a set of TE-PtmP or TE-PtP interfaces, asterisk box should also include such NT-PtP or NT-PtmP interfaces. For instance, would you say that in the UK, most PBXes are using TE-PtP ? You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried the new release). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 08:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] libpri and NT-Point to multi-point Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ael queue gosub already has PBX structure??
2008/12/17 Mark Michelson mmichel...@digium.com
Giedrius Augys wrote:
Hello,
I want that after client and queue member call would be established,
cmd queue runs some 'procedures' . So I am using cmd Queue option
'gosub'. This is my example of ael :
context QUEUE {
_X. = {
Ringing();
Wait(4);
Answer();
Queue(${Queue},wr,,,60,,,check-record);
Hangup();
};
};
macro check-record() {
Set(MEMBERNUMBER=${CUT(MEMBERINTERFACE,@,1)});
Set(MEMBERNUMBER=${CUT(MEMBERNUMBER,/,2)});
return;
};
Everything works normal, but when the client's and queue call
establishes , I get this WARNING:
-- Local/1...@cc-out-da9a;1 answered SIP/xxx.xxx.xx-12d132d0
[Dec 17 20:52:12] WARNING[3849]: pbx.c:3656 __ast_pbx_run:
SIP/sip.call.lt-12d132d0 already has PBX structure??
== Starting SIP/sip.call.lt-12d132d0 at check-record,s,0 failed so
falling back to exten 's'
-- Executing [...@check-record:1] Set(SIP/sip.call.lt-12d132d0,
MEMBERNUMBER=Local/123) in new stack
-- Executing [...@check-record:2] Set(SIP/sip.call.lt-12d132d0,
MEMBERNUMBER=123) in new stack
What I'm missing? Something wrong with ael syntax/structure ?
Thanks in advance
--
Pagarbiai / Best Regards,
Giedrius Augys
This is a bug you are experiencing, which I fixed recently in a series of
commits. Assuming you are using a 1.6 tag, the next build should have this
problem fixed.
Mark Michelson
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Hi,
yes I'm using 1.6.0.1 version.
--
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Installing Asterisk v1.6 on Ubuntu Intrepid?
Scott, I had the same problem when I downloaded http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz This downloaded asterisk-1.6.0.2.tar.gz To fix the problem I downloaded http://downloads.digium.com/pub/asterisk/asterisk-1.6.0.3-rc1.tar.gz and I was able to compile without any problems. James On Tue, Dec 16, 2008 at 5:55 PM, Scott Berry n7...@northlc.com wrote: Hi Tillman, I am havingthe same problem can you expand on your answer here? I am not sure I understand what your saying. Are you saying that this is really not an Asterisk problem? And just another thought. Where is sentinel coming from? Interesting I wounder if it's something left over from another version of Asterisk from an early version? Scott On Tue, 2008-12-16 at 13:38 -0600, Tilghman Lesher wrote: On Tuesday 16 December 2008 13:14:06 Christian wrote: Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c - manager.o manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 In neither the 1.6.0 branch nor the 1.6.1 branch is SENTINEL used within main/manager. So you're clearly using a third party patch. You need to contact the person from whom you obtained that patch and ask this question. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 to AS5400 using H.323 (ooh323) inbound working but outbound doesn't
I have the following setup: DS3 - Cisco AS5400 - H.323 (ooh323) - Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We ran this through the Cisco output interpreter and it says that there may be a problem with the outbound DS3. But we have disproved that already. The other thing that I noticed is the the calling party number Type is set to Unknown on Asterisk but set to National on the CME. I don't see a way to specify the Type in ooh323.conf. At this point I'm just looking for any ideas here. Most of the setups I have googled are Asterisk to AS5400 using SIP. We started doing some debugs on the AS5400. Here is a sample of the AS5400 for the Call Manager Express (the one that works): 993241: Dec 16 14:53:12: //-1//DPM/dpMatchPeersCore: Calling Number=96519590, Called Number=96519590, Peer Info Type=DIALPEER_INFO_SPEECH 993242: Dec 16 14:53:12: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=96519590 993243: Dec 16 14:53:12: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST 993244: Dec 16 14:53:12: //-1//DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=101 993245: Dec 16 14:53:12: //-1/122998F18026/DPM/ dpAssociateIncomingPeerCore: Calling Number=555, Called Number=96519590, Voice- Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH 993246: Dec 16 14:53:12: //-1/122998F18026/DPM/ dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt 993247: Dec 16 14:53:12: //-1/122998F18026/DPM/dpMatchPeersCore: Calling Number=, Called Number=96519590, Peer Info Type=DIALPEER_INFO_SPEECH 993248: Dec 16 14:53:12: //-1/122998F18026/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=96519590 993249: Dec 16 14:53:12: //-1/122998F18026/DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST 993250: Dec 16 14:53:12: //-1/122998F18026/DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=101 993251: Dec 16 14:53:12: ISDN Se7/0:1:23 Q931: Applying typeplan for sw-type 0x2 is 0x2 0x1, Calling num 555 993252: Dec 16 14:53:12: ISDN Se7/0:1:23 Q931: Applying typeplan for sw-type 0x2 is 0x2 0x1, Called num 3146519590 993253: Dec 16 14:53:12: ISDN Se7/0:1:23 Q931: TX - SETUP pd = 8 callref = 0x00F3 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98397 Exclusive, Channel 23 Net Specific Fac i = 0x00E1 Calling Party Number i = 0x2180, '555' Plan:ISDN, Type:National Called Party Number i = 0xA1, '3146519590' Plan:ISDN, Type:National 993254: Dec 16 14:53:12: ISDN Se7/0:1:23 Q931: RX - CALL_PROC pd = 8 callref = 0x80F3 Channel ID i = 0xE9808397 Exclusive, Interface 0, Channel 23 993255: Dec 16 14:53:16: ISDN Se7/0:1:23 Q931: RX - PROGRESS pd = 8 callref = 0x80F3 Progress Ind i = 0x8381 - Call not end-to-end ISDN, may have in-band info 993256: Dec 16 14:53:17: ISDN Se7/0:1:23 Q931: RX - ALERTING pd = 8 callref = 0x80F3 993257: Dec 16 14:53:20: ISDN Se7/0:1:23 Q931: RX - CONNECT pd = 8 callref = 0x80F3 993258: Dec 16 14:53:20: ISDN Se7/0:1:23 Q931: TX - CONNECT_ACK pd = 8 callref = 0x00F3 993259: Dec 16 14:53:20: //1797/122998F18026/SIP/Call/sipSPICallInfo: The Call Setup Information is: Call Control Block (CCB) : 0x63F39424 State of The Call: STATE_ACTIVE TCP Sockets Used : NO Calling Number : 555 Called Number: 96519590 Source IP Address (Sig ): 10.200.90.20 Destn SIP Req Addr:Port : 10.200.11.250:5060 Destn SIP Resp Addr:Port : 10.200.11.250:64816 Destination Name : 10.200.11.250 993260: Dec 16 14:53:20: //1797/122998F18026/SIP/Call/ sipSPIMediaCallInfo: Number of Media Streams: 1 Media Stream : 1 Negotiated Codec : g711ulaw Negotiated Codec Bytes : 160 Negotiated Dtmf-relay: 7 Dtmf-relay Payload : 0 Source IP Address (Media): 10.200.90.20 Source IP Port(Media): 21846 Destn IP Address (Media): 10.200.11.250 Destn IP Port(Media): 16860 Orig Destn IP Address:Port (Media): 0.0.0.0:0 993261: Dec 16 14:53:23: ISDN Se7/0:1:23 Q931: RX - DISCONNECT pd = 8 callref = 0x80F3 Cause i = 0x8090 - Normal call clearing 993262: Dec 16 14:53:23: ISDN Se7/0:1:23 Q931: TX - RELEASE pd = 8 callref = 0x00F3 993263: Dec 16 14:53:23:
Re: [asterisk-users] RDNIS and asterisk
Isn't that the ${exten} number? In other words, the number called.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Tony Mountifield
-- Sent: 17 December 2008 10:17
-- To: asterisk-users@lists.digium.com
-- Subject: Re: [asterisk-users] RDNIS and asterisk
--
-- In article 49483005.8030...@dotr.com,
-- Julian Lyndon-Smith aster...@dotr.com wrote:
-- I have a couple of numbers that are diverted to a number that is
-- conected to an isdn30 card, running asterisk 1.4.
--
-- eg.
--
-- 123456 = 22334455
-- 654321 = 22334455
--
-- What I would like to know is the number of the orginal number
-- dialled
-- (123456 or 654321). I thought that RDNIS was the answer, but it
is
-- always coming up blank.
--
-- When I did a debug on the pri span, I saw the following message
--
-- Unable to handle ROSE operation 15
--
-- is this the cause of my problem ?
--
-- Don't know about that error, but in the pri debug output, did you
see
-- any mention of the originally dialled number, or only the
translated
-- number?
--
-- If the originally dialled number is not presented in an
information
-- element somewhere, then it would be a bit of a challenge for
Asterisk
-- to infer it! :-)
--
-- Just found this message, which seems to refer to the same issue:
--
http://lists.digium.com/pipermail/asterisk-users/2007-July/191858.html
--
-- If your original number does appear in a ROSE IE simlar to that
shown
-- in
-- the above message, then it may be that libpri needs updating to
handle
-- it.
--
-- Can you get a second destination number on the same ISDN30 and
then
-- divert one of the original numbers to that instead?
--
-- Cheers
-- Tony
-- --
-- Tony Mountifield
-- Work: t...@softins.co.uk - http://www.softins.co.uk
-- Play: t...@mountifield.org - http://tony.mountifield.org
--
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Re: [asterisk-users] RDNIS and asterisk
In article 49483005.8030...@dotr.com, Julian Lyndon-Smith aster...@dotr.com wrote: I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 = 22334455 654321 = 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message Unable to handle ROSE operation 15 is this the cause of my problem ? Don't know about that error, but in the pri debug output, did you see any mention of the originally dialled number, or only the translated number? If the originally dialled number is not presented in an information element somewhere, then it would be a bit of a challenge for Asterisk to infer it! :-) Just found this message, which seems to refer to the same issue: http://lists.digium.com/pipermail/asterisk-users/2007-July/191858.html If your original number does appear in a ROSE IE simlar to that shown in the above message, then it may be that libpri needs updating to handle it. Can you get a second destination number on the same ISDN30 and then divert one of the original numbers to that instead? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri and NT-Point to multi-point
2008/12/17 Andrew Thomas a...@datavox.co.uk I have piggy backed a few PBX's off the back of a B410P (4 x BRI) card with no problems. The ones I used for testing were the Avaya IP Office, Siemens Hi-Path/Hi-Com and various old Panasonics. All I had to do was to turn on the 100ohm termination on my S0 ports (set as NT on the B410P of course). I actually have a similar set-up at the moment on our main asterisk system. 2 x BRI trunks (ports 1 2 set as TE ptp - for the DDI's etc) and 2 x BRI S0 bus (ports 3 4 set as NT ptmp) running and ISDN modem on each (good old Fritz! ones). Fine, so, using this setup as an example, would say the norm in the UK, is to connect to ISDN-BRI in ptp (reading from ports 1 and 2 configuration), or to connect using ptmp (reading from ports 3 and 4 configuration, dedicated to isdn modems) ? So it is possible to run ptmp on NT ports using mISDN - just remember to turn on 100ohm termination on the ISDN card if you only have one device per port. HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 09:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] libpri and NT-Point to multi-point Hello Andrew, 2008/12/17 Andrew Thomas a...@datavox.co.uk If you are connecting to BRI lines then you should be TE - not NT. Yes of course, you're right. I was mostly referring to this : ISDN --BRI asterisk -BRI- legacy PBX Then, in this case, as legacy PBX has a set of TE-PtmP or TE-PtP interfaces, asterisk box should also include such NT-PtP or NT-PtmP interfaces. For instance, would you say that in the UK, most PBXes are using TE-PtP ? You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried the new release). HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 17 December 2008 08:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] libpri and NT-Point to multi-point Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK, Germany, Italy, ...) ? Do you think this is needed ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 upgrade issues
It is precisely relevant to this issue. All subroutines, whether they're
called macros or not, in AEL (in 1.6) are Gosub routines. So to invoke that
subroutine, you need to call out with Gosub, not with Macro. So it probably
should be along the lines of: Gosub(outbound,s,1
(${EXTEN},provider1,provider2)).
Thanks to all who replied. Looks like I just need to do a bit of
extensions.conf find/replace then.
Any thoughts on the CDR issue?
TIA.
Regards,
Chris
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[asterisk-users] Install app_rxfax and app_txfax in 1.4 with Lenny
Hi, I've read README file in agx-ast-addons-1.4.17.5.tar.bz2 It says Install libTiff =3.8 and 4.0 Should you really use this agx-ast-addons to get app_rxfax and app-_txfax running with latest 1.4.22 ? If positive, should you take this libtiff warning into account ? If positive, where can you find such libtiff version as Debian repository (I didn't check alternate distrib) includes libtiff4 but no libtiff3 not libtiff. Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to tell when a issue actually gets in a released version
This bug report http://bugs.digium.com/print_bug_page.php?bug_id=12038 apparently has been fixed. I dont see anything on the page saying what released version of asterisk this is in. How can I tell that? jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alcatel OXE + Asterisk as external IVR
On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote: 2008/12/17 Artifex Maximus artife...@gmail.com Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI I've talked with support person at Alcatel and he said that Q.931 cannot handle this situation because after calls leave OXE it does not know anything so I cannot hangup in Asterisk and call will use two channel. Is it right? He said that ABCF2 or Q.SIG is able handling this situation because Q.SIG is an extension to Q.931. I take some search on topic and find out that Asterisk's Q.SIG not fully implemented. Is Asterisk implementation enough for this kind of setup? What is needed is that the Asterisk box should either : - forward incoming call to the right endpoint, using a single channel, - open a second channel and remain in media path till it ends. Thanks for your answer! You are right and first option what I am looking for. I have asked support staff and sending back DTMF on open channel does not help. I'm not an authority on this topic, but I would say that, as OXE and asterisk are connected through an E1/T1 link, - you must upgrade OXE and Asterisk to Q.SIG to get forwarding option (and check asterisk's QSIG supports Call Deflection), - casual PRI is enough if you stick with 2 channels option. Unfortunately I am not expert on this topic as well but second option is not good for us. The question is how good Asterisk's Q.SIG implementation for this task. If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. Often we have more than 15 calls at same time and that is why first option is not acceptable. Bye, Zsolt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WTB: Digium 1 or 4 ports E1 Cards
http://shop.ebay.com/items/_W0QQ_nkwZdigiumQQ_armrsZ1QQ_fromZR40QQ_mdoZ 2008/12/17 Pete Kay pete...@gmail.com Hi, I am looking to buy 2 used 1 or 4 ports E1 Cards. If you have one, would you please contact me? Thanks, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RDNIS and asterisk
The original poster is looking for RDNIS as the number initially dialed, i.e. the DNIS recognized by the first PSTN switch handling the call. The call may have been diverted to a different number, e.g. unconditionally forwarded to a call center (answering service, for us older types). If there are multiple numbers forwarded to the same DNIS at the call center, the RDNIS is required to determine which number (client) was originally dialed. The inability to handle the ROSE operation 15 is the issue. In the example posted by someone (I can't tell who) the information is present, it just isn't available to the application: http://lists.digium.com/pipermail/asterisk-users/2007-July/191858.html This message appears: !! Unable to handle ROSE operation 15 [ 30 19 02 01 01 0A 01 02 A1 11 A0 0F A1 0D 0A 01 02 12 08 32 32 34 35 38 34 30 35 ] - [0..22458405] The number originally dialed in this example was 22458405 (note the hex digits within the message: 32 32 34 35 38 34 30 35) --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
2008/12/17 Barry L. Kline blkl...@attglobal.net Bill Andersen wrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill Not when you take the time to properly trim your reply it's not. BK ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Q: What is the most annoying thing in email? the most annoyng is the guy who wrote ten billions mails saing top posting... top posting David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and NAT one way audio
On Thu, Dec 18, 2008 at 12:46 AM, Silvia Menendez silvia.menen...@gmail.com wrote: Hello may situation is the next: Asterisk -- NAT1 (router)--- internet -- NAT2 (router) -- x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone to the x-lite. From the x-lite to de ip phone there is no audio. I have made port fordwarding in the router 1. I have opened ports 5060-5070 to SIP and 1-2 to RTP. In the sip.conf file I have externip, localnet, bindport, bindaddres, nat = yes etc. I have used wireshark to see the frames and there is rtp traffic in both direccion so i don´t know what the problem is. I have used the x-lite in the same LAN than the ip-phone and all is good so the two phones works fine. What is the problem? Thank you in advance a pointer to check, in rtp.conf just make sure that rtp start port is set explicitly rtpstart=1, cause default rtpstart is 5000 so opening port 1-2 in router without setting this may not help. -- Thanks Regards, Godson Gera http://godson.in/voip-asterisk-consultant-hyderabad-india ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RDNIS and asterisk
If the diversion takes place in asterisk then the dialplan can set a
variable before it diverts and then use that variable at the destination of
the diversion.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Julian Lyndon-Smith aster...@dotr.com
Reply-To: aster...@dotr.com, Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Wed, 17 Dec 2008 11:07:05 +
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] RDNIS and asterisk
No, ${exten} is the final destination number
myphone calls 123456, which is diverted to 22334455 would givc an
${exten} of 22334455, but I wanted to know the 123456.
Julian
Andrew Thomas wrote:
Isn't that the ${exten} number? In other words, the number called.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Tony Mountifield
-- Sent: 17 December 2008 10:17
-- To: asterisk-users@lists.digium.com
-- Subject: Re: [asterisk-users] RDNIS and asterisk
--
-- In article 49483005.8030...@dotr.com,
-- Julian Lyndon-Smith aster...@dotr.com wrote:
-- I have a couple of numbers that are diverted to a number that is
-- conected to an isdn30 card, running asterisk 1.4.
--
-- eg.
--
-- 123456 = 22334455
-- 654321 = 22334455
--
-- What I would like to know is the number of the orginal number
-- dialled
-- (123456 or 654321). I thought that RDNIS was the answer, but it
is
-- always coming up blank.
--
-- When I did a debug on the pri span, I saw the following message
--
-- Unable to handle ROSE operation 15
--
-- is this the cause of my problem ?
--
-- Don't know about that error, but in the pri debug output, did you
see
-- any mention of the originally dialled number, or only the
translated
-- number?
--
-- If the originally dialled number is not presented in an
information
-- element somewhere, then it would be a bit of a challenge for
Asterisk
-- to infer it! :-)
--
-- Just found this message, which seems to refer to the same issue:
--
http://lists.digium.com/pipermail/asterisk-users/2007-July/191858.html
--
-- If your original number does appear in a ROSE IE simlar to that
shown
-- in
-- the above message, then it may be that libpri needs updating to
handle
-- it.
--
-- Can you get a second destination number on the same ISDN30 and
then
-- divert one of the original numbers to that instead?
--
-- Cheers
-- Tony
-- --
-- Tony Mountifield
-- Work: t...@softins.co.uk - http://www.softins.co.uk
-- Play: t...@mountifield.org - http://tony.mountifield.org
--
-- ___
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Re: [asterisk-users] Alcatel OXE + Asterisk as external IVR
2008/12/17 Artifex Maximus artife...@gmail.com Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR transfer back to specified Pilot in OXE with Dial (using two channels) 6. Asterisk hangup (free both channels) 7. OXE connect the PSTN incoming line with Pilot as extension transfer does I've talked with support person at Alcatel and he said that Q.931 cannot handle this situation because after calls leave OXE it does not know anything so I cannot hangup in Asterisk and call will use two channel. Is it right? He said that ABCF2 or Q.SIG is able handling this situation because Q.SIG is an extension to Q.931. I take some search on topic and find out that Asterisk's Q.SIG not fully implemented. Is Asterisk implementation enough for this kind of setup? I am using Asterisk 1.6.0.3-rc1 with dahdi-*-2.1.0 on Ubuntu Server 8.10. Thanks, Zsolt Hi, What is needed is that the Asterisk box should either : - forward incoming call to the right endpoint, using a single channel, - open a second channel and remain in media path till it ends. I'm not an authority on this topic, but I would say that, as OXE and asterisk are connected through an E1/T1 link, - you must upgrade OXE and Asterisk to Q.SIG to get forwarding option (and check asterisk's QSIG supports Call Deflection), - casual PRI is enough if you stick with 2 channels option. If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet size limit for HDLC?
Hi, I figured out, that app_pppd suffered from overruns under high out traffic. (ping -s 600 destip was already high in this context.) I've just made a quick and dirty hack to fix it. If interested, just download the original package by Sirrix (as mentioned on VoIP-Wiki) and the replace their app_ppp.c by: http://planinternet.net/download/voip/asterisk/app_pppd.c Maybe I will later find the time to bundle a complete package, like the one by Sirrix. Regards, Roger. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
On Wed, Dec 17, 2008 at 07:03:15PM -0200, David fire wrote: 2008/12/17 Barry L. Kline blkl...@attglobal.net Bill Andersen wrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill Not when you take the time to properly trim your reply it's not. Q: What is the most annoying thing in email? the most annoyng is the guy who wrote ten billions mails saing top posting... top posting Top posting is a very important policy issue for this list. We all need to be constantly reminded about its importance and hence it is good to see you guys keeping us reminded about it. If you don't agree with me, simply don't reply :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Alcatel OXE + Asterisk as external IVR
Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR transfer back to specified Pilot in OXE with Dial (using two channels) 6. Asterisk hangup (free both channels) 7. OXE connect the PSTN incoming line with Pilot as extension transfer does I've talked with support person at Alcatel and he said that Q.931 cannot handle this situation because after calls leave OXE it does not know anything so I cannot hangup in Asterisk and call will use two channel. Is it right? He said that ABCF2 or Q.SIG is able handling this situation because Q.SIG is an extension to Q.931. I take some search on topic and find out that Asterisk's Q.SIG not fully implemented. Is Asterisk implementation enough for this kind of setup? I am using Asterisk 1.6.0.3-rc1 with dahdi-*-2.1.0 on Ubuntu Server 8.10. Thanks, Zsolt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
Barry L. Kline wrote: Bill Andersen wrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill Not when you take the time to properly trim your reply it's not. BK To me top posting is like people talking about SIP Trunks. There is no such thing as a SIP Trunk. There are SIP connections, peers, friends, etc. The term is simply a marketing buzzword to make people that don't know much about VoIP feel all warm and fuzzy about a product. You're not going to be able to make people stop top posting and I'm not going to be able to make people stop using wrong or misleading terms like SIP Trunk. If you try all you are going to do is piss people off and stress yourself out. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
On 12/17/2008 Eric ManxPower Wieling wrote: You're not going to be able to make people stop top posting and I'm not going to be able to make people stop using wrong or misleading terms like SIP Trunk. If you try all you are going to do is piss people off and stress yourself out. Why this? For me it is simple. When top posting or a messy top /bottom mix makes is difficult for me to read or understand, I simply ignore that mail and open the next one. I do so since I feel that the poster did not bother to pay me the slightest respect to make his mail as readable as possible for people in order to get the highest possible quota of answers from readers. There are always other people whose questions deserve to be answered. ;-) Kind regards Eberhard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 to AS5400 using H.323 (ooh323) inbound working but outbound doesn't
You can do a translation rule on the outbound peer, like
voice translation-rule 10
rule 1 /.*/ /\0/ type any national plan any isdn
voice translation-profile SET_TypePlan
translate calling 10 {or} translate called 10 (whatweven you want to
change)
and in the DS3 trunk if you have a trunk group created
trunk group OutTrunkGroupDS3
translation-profile outgoing SET_TypePlan
LTenorio
Ray Seals wrote:
I have the following setup: DS3 - Cisco AS5400 - H.323 (ooh323) -
Asterisk
Inbound calls work great but outbound calls fail. So to check and
make sure we have outbound calling ability on the DS3 we setup a Cisco
Call Manager Express and it can make outbound calls both local and
long distance with no problems.
The failure code is Cause i = 0x8381 - Unallocated/unassigned number.
We ran this through the Cisco output interpreter and it says that
there may be a problem with the outbound DS3. But we have disproved
that already.
The other thing that I noticed is the the calling party number Type is
set to Unknown on Asterisk but set to National on the CME. I don't
see a way to specify the Type in ooh323.conf. At this point I'm just
looking for any ideas here. Most of the setups I have googled are
Asterisk to AS5400 using SIP.
We started doing some debugs on the AS5400. Here is a sample of the
AS5400 for the Call Manager Express (the one that works):
993241: Dec 16 14:53:12: //-1//DPM/dpMatchPeersCore:
Calling Number=96519590, Called Number=96519590, Peer Info
Type=DIALPEER_INFO_SPEECH
993242: Dec 16 14:53:12: //-1//DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=96519590
993243: Dec 16 14:53:12: //-1//DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
993244: Dec 16 14:53:12: //-1//DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=101
993245: Dec 16 14:53:12: //-1/122998F18026/DPM/
dpAssociateIncomingPeerCore:
Calling Number=555, Called Number=96519590, Voice-
Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
993246: Dec 16 14:53:12: //-1/122998F18026/DPM/
dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
993247: Dec 16 14:53:12: //-1/122998F18026/DPM/dpMatchPeersCore:
Calling Number=, Called Number=96519590, Peer Info
Type=DIALPEER_INFO_SPEECH
993248: Dec 16 14:53:12: //-1/122998F18026/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=96519590
993249: Dec 16 14:53:12: //-1/122998F18026/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
993250: Dec 16 14:53:12: //-1/122998F18026/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=101
993251: Dec 16 14:53:12: ISDN Se7/0:1:23 Q931: Applying typeplan for
sw-type 0x2 is 0x2 0x1, Calling num 555
993252: Dec 16 14:53:12: ISDN Se7/0:1:23 Q931: Applying typeplan for
sw-type 0x2 is 0x2 0x1, Called num 3146519590
993253: Dec 16 14:53:12: ISDN Se7/0:1:23 Q931: TX - SETUP pd = 8
callref = 0x00F3
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98397
Exclusive, Channel 23
Net Specific Fac i = 0x00E1
Calling Party Number i = 0x2180, '555'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '3146519590'
Plan:ISDN, Type:National
993254: Dec 16 14:53:12: ISDN Se7/0:1:23 Q931: RX - CALL_PROC pd = 8
callref = 0x80F3
Channel ID i = 0xE9808397
Exclusive, Interface 0, Channel 23
993255: Dec 16 14:53:16: ISDN Se7/0:1:23 Q931: RX - PROGRESS pd = 8
callref = 0x80F3
Progress Ind i = 0x8381 - Call not end-to-end ISDN, may have
in-band info
993256: Dec 16 14:53:17: ISDN Se7/0:1:23 Q931: RX - ALERTING pd = 8
callref = 0x80F3
993257: Dec 16 14:53:20: ISDN Se7/0:1:23 Q931: RX - CONNECT pd = 8
callref = 0x80F3
993258: Dec 16 14:53:20: ISDN Se7/0:1:23 Q931: TX - CONNECT_ACK pd =
8 callref = 0x00F3
993259: Dec 16 14:53:20: //1797/122998F18026/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x63F39424
State of The Call: STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 555
Called Number: 96519590
Source IP Address (Sig ): 10.200.90.20
Destn SIP Req Addr:Port : 10.200.11.250:5060 http://10.200.11.250:5060
Destn SIP Resp Addr:Port : 10.200.11.250:64816
http://10.200.11.250:64816
Destination Name : 10.200.11.250
993260: Dec 16 14:53:20: //1797/122998F18026/SIP/Call/
sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Negotiated Dtmf-relay: 7
Dtmf-relay
Re: [asterisk-users] top posting again [was: Re: CDR Design]
OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M @#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT MSN IS FOR. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eberhard Roloff Sent: Wednesday, December 17, 2008 4:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] On 12/17/2008 Eric ManxPower Wieling wrote: You're not going to be able to make people stop top posting and I'm not going to be able to make people stop using wrong or misleading terms like SIP Trunk. If you try all you are going to do is piss people off and stress yourself out. Why this? For me it is simple. When top posting or a messy top /bottom mix makes is difficult for me to read or understand, I simply ignore that mail and open the next one. I do so since I feel that the poster did not bother to pay me the slightest respect to make his mail as readable as possible for people in order to get the highest possible quota of answers from readers. There are always other people whose questions deserve to be answered. ;-) Kind regards Eberhard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
On Wed, Dec 17, 2008 at 8:39 PM, Bill Andersen ander...@mwdental.comwrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! ./bill Posting either way can be good, bad, or ugly. The key is consistency. What I find most annoying is a mix of top and bottom posting within a single mailing list or especially withing a single thread. Mailing lists in general have adopted the old Usenet convention of bottom posting as a standard. The Usenet idea, as I understand it, is that you never knew at what point someone would begin reading a thread. Slow servers, missed messages, and limited retention policies meant that a reader might not see the beginning of a thread or could miss parts of a thread. If you always bottom post, and snip accordingly, then the topic, context, and conversation happen in natural top to bottom order even if parts or history is missed. I personally prefer top posting in direct emails where the context of the conversation is already in my mind. For mailing lists and Usenet I prefer to bottom post so that it remains consistent and because they are often archived publicly. I appreciate an archive where I can follow the flow of conversation from top to bottom and know that I have reached a conclusion of discussion. Sadly the new standard seems to be do whatever you want which includes breaking the existing convention on an already started thread such that parts are top posted, parts are bottom posted, and none of the conversation is snipped. -Justin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! I am strongly against anyone posting anything with their bottom. later, PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
Paul Hales wrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! I am strongly against anyone posting anything with their bottom. How about this, don't quote anything. That will fix it until people bitch that they can't follow the threat. I already ignore any top posted threads for the most part, the natural order if older first unless you use Outlook or Outlook Express. If someone is using Outlook Express for e-mail I know they are a moron already and can safely ignore the ignoramus. Look at the headers, you'll see the trend. It's not 100% true all the time, but a good 90%. -Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
On Wed, 17 Dec 2008, Danny Nicholas wrote: OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M @#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT MSN IS FOR. Spoken like a true top-poster... Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record CMD
On Tuesday 16 December 2008 14:51:47 Barton Fisher wrote:
- Original Message -
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 16, 2008 10:40 AM
Subject: Re: [asterisk-users] Record CMD
On Monday 15 December 2008 18:37:05 Barton Fisher wrote:
I don't see a method to detect the success or failure for the Record
CMD.
I'd like to know the reason why the recording ended
Am I wrong?
exten = recordmsg,1,Noop()
exten = recordmsg,n,Record(${NEWPHRASEID}:ulaw|4|180)
So you'd be looking for a RECORD_STATUS, perhaps of SILENCE, MAXLENGTH,
or POUNDKEY, right? That sounds like a reasonable request.
Exactly! but sadly these variables don't seem to exists as far as I can
tell
The point is that you're the first person to make this request. If nobody had
the idea to do it before you, that is precisely the reason it never got done.
Now that it has been requested, it is in queue for trunk and will be in the
next 1.6 to be branched (probably 1.6.2).
--
Tilghman
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Re: [asterisk-users] top posting again [was: Re: CDR Design]
RE Kushner List Account wrote: Paul Hales wrote: In the order in which people normally read text they don't repeat the entire conversation from the beginning each time a question is asked either... Bottom posting is just as bad! I am strongly against anyone posting anything with their bottom. How about this, don't quote anything. That will fix it until people bitch that they can't follow the threat. I already ignore any top posted threads for the most part, the natural order if older first unless you use Outlook or Outlook Express. If someone is using Outlook Express for e-mail I know they are a moron already and can safely ignore the ignoramus. Look at the headers, you'll see the trend. It's not 100% true all the time, but a good 90%. Looks like someone out there isn't happy so I'm going to repost How about this, don't quote anything. That will fix it until people FEMALE DOG that they can't follow the threat. I already ignore any top posted threads for the most part, the natural order is older first unless you use Outlook or Outlook Express. If someone is using Outlook Express for e-mail I know they are a moron already and can safely ignore the ignoramus. Look at the headers, you'll see the trend. It's not 100% true all the time, but a good 90%. -Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
Steve Edwards wrote: On Wed, 17 Dec 2008, Danny Nicholas wrote: OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M @#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT MSN IS FOR. Spoken like a true top-poster... Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Top posting. Bottom posting. Honestly, if you can't use an effing scrollbar, please tell me so I can take you out back and beat you to death with something heavy. The .5 seconds it takes to scroll from one end of a message to another is no excuse for spending 2 minutes writing a tirade about how you don't like to spend that extra .5 seconds. I swear. You people need to get up, walk away from the computer, go outside and realise that this level of egocentrism is incredibly unhealthy. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record CMD
Exactly! but sadly these variables don't seem to exists as far as I can tell The point is that you're the first person to make this request. If nobody had the idea to do it before you, that is precisely the reason it never got done. Now that it has been requested, it is in queue for trunk and will be in the next 1.6 to be branched (probably 1.6.2). -- Tilghman Could it be back-ported to 1.4? Really not ready for 1.6 Thanks, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
On December 17, 2008 05:03:00 pm Eric ManxPower Wieling wrote: To me top posting is like people talking about SIP Trunks. There is no such thing as a SIP Trunk. There are SIP connections, peers, friends, etc. The term is simply a marketing buzzword to make people that don't know much about VoIP feel all warm and fuzzy about a product. I thought the term SIP trunk came from old PBX-heads trying to apply the term SIP to a destination route, much like LD trunks, POTS trunks and even remote office trunks. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
you are soamming my mail box whit this useless discution the solution is doble posting (top and bottom) 2008/12/17 Andrew Kohlsmith (lists) akli...@mixdown.ca On December 17, 2008 05:03:00 pm Eric ManxPower Wieling wrote: To me top posting is like people talking about SIP Trunks. There is no such thing as a SIP Trunk. There are SIP connections, peers, friends, etc. The term is simply a marketing buzzword to make people that don't know much about VoIP feel all warm and fuzzy about a product. I thought the term SIP trunk came from old PBX-heads trying to apply the term SIP to a destination route, much like LD trunks, POTS trunks and even remote office trunks. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users you are soamming my mail box whit this useless discution the solution is doble posting (top and bottom) -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
Everyone read this top down for your IVR wav file. Press 9 for the company directory Press 8 for the billing department Press 1 for technical support Press 0 for the operator Next let us know who calls into your PBX complaining that your menu is whacked. Now discussing PBX related issues, that is on topic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SIP Sent: Wednesday, December 17, 2008 15:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] Steve Edwards wrote: Top posting. Bottom posting. Honestly, if you can't use an effing scrollbar, please tell me so I can take you out back and beat you to death with something heavy. The .5 seconds it takes to scroll from one end of a message to another is no excuse for spending 2 minutes writing a tirade about how you don't like to spend that extra .5 seconds. I swear. You people need to get up, walk away from the computer, go outside and realise that this level of egocentrism is incredibly unhealthy. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
On December 17, 2008 06:59:19 pm David fire wrote: you are soamming my mail box whit this useless discution the solution is doble posting (top and bottom) It's a public mailing list. If you're having trouble managing it, you may want to try a digest version, or perhaps a moderated list. -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
On Wed, 17 Dec 2008, Darryl Dunkin wrote: Steve Edwards wrote: Top posting. Bottom posting. Honestly, if you can't use an effing scrollbar, please tell me so I can take you out back and beat you to death with something heavy. The .5 seconds it takes to scroll from one end of a message to another is no excuse for spending 2 minutes writing a tirade about how you don't like to spend that extra .5 seconds. I swear. You people need to get up, walk away from the computer, go outside and realise that this level of egocentrism is incredibly unhealthy. N. I != N. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] user entry as variables
I want to take series of user entered (via phone keypad) options/numeric entry fields and use these with an AGI script. I have looked through voip-info and I can't find any Asterisk functions specifically for this. Any guidance please? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AGX addons compile issues
Has anyone seen this before, and know what is happening? u...@host:~/asterisk/agx-ast-addons# ./build.sh -- Configuring done -- Generating done -- Build files have been written to: /root/asterisk/agx-ast-addons [ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o Linking C shared module dist/app_devstate.so [ 11%] Built target app_devstate [ 22%] Building C object CMakeFiles/app_nv_backgrounddetect.dir/app_nv_backgrounddetect.o Linking C shared module dist/app_nv_backgrounddetect.so [ 22%] Built target app_nv_backgrounddetect [ 33%] Building C object CMakeFiles/app_nv_faxdetect.dir/app_nv_faxdetect.o Linking C shared module dist/app_nv_faxdetect.so [ 33%] Built target app_nv_faxdetect [ 44%] Building C object CMakeFiles/app_pickup2.dir/app_pickup2.o Linking C shared module dist/app_pickup2.so [ 44%] Built target app_pickup2 [ 55%] Building C object CMakeFiles/app_rxfax.dir/app_rxfax.o cc1: warnings being treated as errors /root/asterisk/agx-ast-addons/app_rxfax.c: In function 'phase_e_handler': /root/asterisk/agx-ast-addons/app_rxfax.c:126: warning: implicit declaration of function 't30_get_local_ident' /root/asterisk/agx-ast-addons/app_rxfax.c:127: warning: implicit declaration of function 't30_get_far_ident' /root/asterisk/agx-ast-addons/app_rxfax.c: In function 'rxfax_exec': /root/asterisk/agx-ast-addons/app_rxfax.c:380: warning: implicit declaration of function 't30_set_local_ident' /root/asterisk/agx-ast-addons/app_rxfax.c:383: warning: implicit declaration of function 't30_set_header_info' /root/asterisk/agx-ast-addons/app_rxfax.c:385: warning: passing argument 2 of 't30_set_phase_b_handler' from incompatible pointer type /root/asterisk/agx-ast-addons/app_rxfax.c:386: warning: passing argument 2 of 't30_set_phase_d_handler' from incompatible pointer type make[2]: *** [CMakeFiles/app_rxfax.dir/app_rxfax.o] Error 1 make[1]: *** [CMakeFiles/app_rxfax.dir/all] Error 2 make: *** [all] Error 2 u...@host:~/asterisk/agx-ast-addons# ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNOW-1.5.0-beta1 Installation Error
I am getting the following error during AsteriskNow installation I am using the following AsteriskNOW-1.5.0-beta1-i386-1of1.iso Here is the error I could piece together as I don't have access to the screen: EIP: [c041041c] powernow8k_init Kernel panic - not syncing: Fatal exception The machine is an old PII. Windows 2000 was previously on the system and worked. Other than that I don't have too much more information. I am going to attempt and install AsteriskNOW-1.0.2.1-x86-disc1.iso to see if I have the same problem. I am a Windows developer so I am still hacking my way around linux in general. Please be patient and clear w/o assuming too much J. Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] user entry as variables
On Wed, 17 Dec 2008, Michael wrote: I want to take series of user entered (via phone keypad) options/numeric entry fields and use these with an AGI script. I have looked through voip-info and I can't find any Asterisk functions specifically for this. Try show agi (1.2) or agi show (1.4) at the Asterisk console. You can collect the digits in your dialplan and stuff them into channel variables which you can retrieve in your AGI using the AGI command GET VARIABLE or you can use AGI commands like GET DATA. It kind of depends on what you are doing. I use GET VARIABLE to retrieve the credit card number and expiration date from channel variables. I use GET DATA to walk the user through a custom bulletin board system. Try show agi get data or agi show get data. FWIW, I'm a big fan of writing AGIs in C. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
