Re: RE: Re: [asterisk-users] setting call-limits
Hi, > I believe you need to setup hints for call-limit to work. can you explain? I don't find any information on it... is this a tool or a library? Thanks -- "Feel free" – 10 GB Mailbox, 100 FreeSMS/Monat ... Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] X100P clone not working
Hi Franck, NOACPI and the sound must be more clear. And, of course, have you tell to /usr/src/zaptel/zconfig.h and /usr/src/asterisk/Makefil what kind of processor you have and enabled MMX if possible before to compile ? Good Luck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Frank Darner Envoyé : dimanche 23 juillet 2006 23:41 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] X100P clone not working Am Sunday 23 July 2006 20:58 schrieb Walter Willis: > look udev rules??? the problem was related that ztcfg did not find zaptel.com -c /etc/asterisk/zaptel.conf has solved this issue #ztcfg --help -c -- Use instead of /etc/zaptel.conf my failure, I should read man page more carefully now am trying to get it working, the sound is still unreliable ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missing close quote in CallerID breaks SIP. . .workaround?
I posted about this some while back, and at that point was told "the
remote end is broken, nothing we can do about it."
The problem: for whatever reason, some CallerID names come in broken.
There is an example CLI trace shown below.
My question: is there anything I can do to fix this, since there's
nothing I can really do about the broken value being passed in? The
call never completes in this instance. . .
Thanks.
B.
** snip **
Jul 24 01:02:10 WARNING[180]: chan_sip.c:1559 get_in_brackets: No
closing quote found in '"Lubbock T
;tag=f6ae058c3893f37fo1'
Jul 24 01:02:10 NOTICE[180]: chan_sip.c:7112 check_user_full: From
address missing 'sip:', using it anyway
Jul 24 01:02:10 WARNING[180]: chan_sip.c:1559 get_in_brackets: No
closing quote found in '"Lubbock T
;tag=f6ae058c3893f37fo1'
Jul 24 01:02:10 WARNING[180]: chan_sip.c:6650 get_destination: Huh? Not
a SIP header ("Lubbock T
;tag=f6ae058c3893f37fo1)?
Jul 24 01:02:30 WARNING[180]: chan_sip.c:1217 retrans_pkt: Maximum
retries exceeded on transmission [EMAIL PROTECTED] for
seqno 101 (Critical Response)
Jul 24 01:02:45 WARNING[180]: chan_sip.c:1217 retrans_pkt: Maximum
retries exceeded on transmission [EMAIL PROTECTED] for
seqno 101 (Critical Response)
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[asterisk-users] MeetMe in Realtime
Gents, does anyone have a conformation about meetme working well in with ARA? I found this particular fix put in somewhere around jan'06 http://bugs.digium.com/view.php?id=5702 Sounds interesting, but not clear from the status if this is actually been merged in newer releases or "safe" to apply to say 1.2.9.1, 1.2.10 or even 1.2.7? Does anyone know? Also, do people have any opinions about using app_meetme over app_conference w/VICIDIAL or vice-versa? Sounds like app_conference is more efficient but doesn't sound it's too stable or mature as much as meetme. That's just what I could gather from some posts on the internet. I could be dead wrong. Any comments are welcome :) Cheers \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
i have been wondering about how the useragents work since a month or two. i have tried every document possible...could not find the answer. if anyone could tell me about the useragents, how they work, what are the factors that are considered while choosing a UA, what makes a particular UA best, it would be of great help. Thanks & Regards Ramya Murthy ph-no- 9845025859 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about asterisk DB
Do you mean the patch can use to replace asterisk DB by ARA? On 7/24/06, Matt Riddell (NZ) <[EMAIL PROTECTED]> wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 RR wrote: > Unplug, I'm sure there are other people with better ideas but if you > see on sineapps, I remember someone having written a patch which > seperates out the the sip registry into a new table. If this is stable Save you searching: http://www.sineapps.com/news.php?rssid=1364 :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFExBN+S6d5vy0jeVcRAgupAJ9b7PVh5bqXX8P232vM/pUTpj2/xgCfQ5aD XqOTQb44gbxSHLxG6G0suFY= =vnEr -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk autoloading of card modules
Hi everyone, I am using Asterisk on CentOS 4.3 with a TDM400P and have managed to get things up and running except this one part. My /etc/sysconfig/zaptel configuration has only one MODULES directive enabled MODULES="$MODULES wctdm" However when I start asterisk it loads the wct1xxp module. Which configuration file controls the loading of card modules? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and H.323
I have been scouring the net the last couple of days looking for some kind of tutorial or walkthrough on setting up a h.323 channel in asterisk. What I need to do is basically this: I have a client who wants to be able to connect to me via h.323 and make a local phone call (local to me, he is in a different country). The call is an automated process and no callee interaction is required. My client simply wants to be able to call a user and give them a verification number and then hang up. He's using some in-house software so unfortunately, h.323 is his only option. Can someone point me to a doc or perhaps give me a simple breakdown of what I need to add to asterisk in order to be able to do this? I am on a tight deadline and my searches have not revealed the information I am looking for. I have built chan_h323 and it is loaded but I'm not sure how to set it up beyond that. Any help would be much appreciated. Thank you Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Negotiation
On Fri July 21 2006 18:33, "Woodoo People .pGa!" <[EMAIL PROTECTED]> wrote: > don't forget the following: > if canreinvite=yes, asterisk will NOT stay in mediapath, so, it going to > ask both parties to negotiate codec, and say hello to the stream. (if > both parties supports g729, and can negotiate it, no licence will be > used) if canreinvite=no, * will STAY in mediapath, so both parties will > negotiate with asterisk itself, and will not care about other side. that > means, if caller has g729, and callee has g711, asterisk WILL transcode. > if both parties have g729, asterisk will NOT transcode, but 2 licence > will be used! Hi there. In your last example, why would any g729 licenses be used? If both parties use g729, wouldn't the call just pass through Asterisk without any licenses being used? Cheers, -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about asterisk DB
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 RR wrote: > Unplug, I'm sure there are other people with better ideas but if you > see on sineapps, I remember someone having written a patch which > seperates out the the sip registry into a new table. If this is stable Save you searching: http://www.sineapps.com/news.php?rssid=1364 :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFExBN+S6d5vy0jeVcRAgupAJ9b7PVh5bqXX8P232vM/pUTpj2/xgCfQ5aD XqOTQb44gbxSHLxG6G0suFY= =vnEr -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with freePBX and Fax reception
Hi have Asterisk running just fine with a single POTS line and a VOIP
line. Recently I have needed to receive some faxes so I've installed
iaxmodem and HylaFax, both of which are working fine.
In freePBX I've configured extension 3999 to point to the iaxmodem
connection. This works just fine if I connect a computer to the FXS
port and send a fax to x3999, the fax is received and handled just
wonderfully.
Next I configured freePBX to direct calls to IAX/3999 when a fax is
detected.
When I try to send a fax from outside it seems to detect that it is a
fax just fine but does not actually dial the fax extension, just deliver
a ring to the caller.
The only doc I could find on this problem is the link below but it seems
to be stuck at the same point.
http://www.aussievoip.com/wiki/index.php?page=freePBX-HylaFax
Any thoughts or ideas about how to get freePBX to dial the fax
extension? In the console output below I am calling out from a FXS port
(Zap/2-1) via the VOIP line back to the FXO port (Zap/1-1) and
presumably to the fax device. But the fax device (IAX2/3999) never gets
called.
If there is an active freePBX forum or mailing list somewhere I would be
happy to ask there as the forum on sourceforge seems to have lots of
questions but no discussion or answers happening.
As an aside, I am wondering if using freePBX was a good idea. It did
make the initial configuration easier but it does seem to have lots of
quirks such as this.
Mike
[EMAIL PROTECTED] etc]# asterisk -r -vvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <[EMAIL PROTECTED]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'show license' for details.
=
Connected to Asterisk 1.2.9.1 currently running on lenovo (pid = 8189)
Verbosity was 3 and is now 7
-- Starting simple switch on 'Zap/2-1'
-- Executing Macro("Zap/2-1", "dialout-trunk|2|1xx||") in
new stack
-- Executing Set("Zap/2-1", "DIAL_TRUNK=2") in new stack
-- Executing Set("Zap/2-1", "DIAL_NUMBER=1xx") in new stack
-- Executing Set("Zap/2-1", "ROUTE_PASSWD=") in new stack
-- Executing GotoIf("Zap/2-1", "1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing Set("Zap/2-1", "GROUP()=OUT_2") in new stack
-- Executing Macro("Zap/2-1", "user-callerid") in new stack
-- Executing GotoIf("Zap/2-1", "0?report") in new stack
-- Executing GotoIf("Zap/2-1", "0?start") in new stack
-- Executing Set("Zap/2-1", "REALCALLERIDNUM=6002") in new stack
-- Executing NoOp("Zap/2-1", "REALCALLERIDNUM is 6002") in new stack
-- Executing Set("Zap/2-1", "AMPUSER=") in new stack
-- Executing Set("Zap/2-1", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("Zap/2-1", "1?report") in new stack
-- Goto (macro-user-callerid,s,9)
-- Executing NoOp("Zap/2-1", "Using CallerID "Channel 2" <6002>") in
new stack
-- Executing Macro("Zap/2-1", "record-enable|6002|OUT") in new stack
-- Executing GotoIf("Zap/2-1", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("Zap/2-1", "recordingcheck|20060723-185730|
1153695447.0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060723-185730|1153695447.0: No AMPUSER db entry for
6002. Not recording
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("Zap/2-1", "No recording needed") in new stack
-- Executing Macro("Zap/2-1", "outbound-callerid|2") in new stack
-- Executing GotoIf("Zap/2-1", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("Zap/2-1", "REALCALLERIDNUM is 6002") in new stack
-- Executing Set("Zap/2-1", "USEROUTCID=") in new stack
-- Executing Set("Zap/2-1", "EMERGENCYCID=") in new stack
-- Executing Set("Zap/2-1", "TRUNKOUTCID="M Hockings" ")
in new stack
-- Executing GotoIf("Zap/2-1", "1?trunkcid") in new stac
Re: [asterisk-users] X100P clone not working
Am Sunday 23 July 2006 20:58 schrieb Walter Willis: > look udev rules??? the problem was related that ztcfg did not find zaptel.com -c /etc/asterisk/zaptel.conf has solved this issue #ztcfg --help -c -- Use instead of /etc/zaptel.conf my failure, I should read man page more carefully now am trying to get it working, the sound is still unreliable ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Operator Console(s)/Shared Call Appearances
Thanks Sebastian - You're right - I have limited experience in this area :) I think the idea below is workable, except we actually want it to work in the other direction - sort of. Essentially we want the receptionist to screen the calls when she's available. The executive should have option to answer the phone if its after hours, or they know the receptionist isn't available (or perhaps they recognize the caller ID and just want to take the call). Can you think of how this might work? I suppose the executive could be a member of his own queue? What do you think about this idea; 1. Call comes in at one of the executive numbers. 2. Executive phone starts ringing for a predetermined time. 3. The callerid is changed to also reflect the name/number of called executive, so that the receptionist knows for who the call was. 4. The call is dropped into a queue for the receptionist (queue because multiple calls to the receptionist at the same time are possible). This setup isn't all that hard, and doesn't require more than 4 sip accounts / phones and one queue, with one agent. Furthermore, if your company starts to grow, and more receptionists that have to answer the phone are needed, it's quite easy, all you have to do is add a sip account, one agent and add that agent to the existing queue. (About 2 minutes...) -- Sebastian Berm iPronto Communications ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error in ubuntu dapper
El vie, 21-07-2006 a las 18:53 -0400, Russell Bryant escribió: > On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote: > > Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to > > bind to 10.152.58.9:5060: Address already in use > > It looks like another application on your system is using port 5060. > Did you install any new software such as a soft phone? > > If you are now using another application that wants to use port 5060, > you will need to configure one of them to use a different port. But, at this point, is the issue worth a bug? I think asterisk should detect the unavailability of the port, and stop with an error message. What can an asterisk running this way help? -- Buon Cammino! don Paolo Benvenuto Vuoi sapere di più su quello che succede qui? leggi il mio diario a http://www.chiesamissionaria.it/diario Visita l'enciclopedia libera, dove puoi contribuire anche tu: http://it.wikipedia.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P clone not working
look udev rules???On 7/23/06, Frank Darner <[EMAIL PROTECTED]> wrote: Tom Lynn:> perhaps not what you're looking for, but reading thru your config, it looks> like you've mis-spelled 'echo cancel' as 'echo cancle'you are right, typothank you___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect XLite with another public IP?
>Now, we have installed "Asterisk" and using for International dialing with Second connection. Now, I have installed "XLite" softphone in our staff systems. I tried >to connect our XLite with our Asterisk server. But, our XLite softphone is unable to connect with Asterisk server. I have given Asterisk server public IP in my >XLite Domain field. But, it is not connecting and is giving an error i.e., " Registration error: 408 - Request timedout". I tried using firewall and without using >firewall. Please tell me how to configure my XLite softphone to connect with my Asterisk server (With other public IP)? Hello. I am very new to this, but I realize that x-Lite hasn't much to configure apart from username, authentication username (which is the same as username), password and domain / proxy (your public * here). Are you behind a dsl (or the like) router? do you have in your workstations assigned IP by dhcp? I would share a little experience. At first time, I installed asterisk with freePBX, and I wasn't able to configure my softphone (x-lite) to connect to *, and outside my lan (yes, with a dlink dsl router in dmz) I was able to connect to it. So, in order to start testing with a clean install, I just removed everything and reinstalled asterisk without anything else, and I can connect now. Doing some search, the nat=[yes|no] externip, localnet parameters apprears. I dont remember exactly what I had configured in my freePBX install to make comparisions, but you should (I guess) configure your sip or iax useras with: nat=yeshost=dynamic canreinvite=no A timeout means that you can't even connect to asterisk, which could mean a network problem (most probably). You could try to connect to asterisk from outside your network, if that works, then your problem is with nat or firewall. You can enter the * console with 'asterisk -rvvv' to see what happens when you try to connect. I hope it helps. Regards,Pablo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to connect XLite with another public IP?
Hi Friends,We have two internet connections (lines) from two Internet Service Providers in our office. So, we have two public IP addresses. We are using one connection for our LAN and to providing internet to our office staff. We are not using second connection. Now, we have installed "Asterisk" and using for International dialing with Second connection. Now, I have installed "XLite" softphone in our staff systems. I tried to connect our XLite with our Asterisk server. But, our XLite softphone is unable to connect with Asterisk server. I have given Asterisk server public IP in my XLite Domain field. But, it is not connecting and is giving an error i.e., " Registration error: 408 - Request timedout". I tried using firewall and without using firewall. Please tell me how to configure my XLite softphone to connect with my Asterisk server (With other public IP)?This is very urgent. Looking forward to your kind response. Thank you.Regards,Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Solved: NAT and externip problem or bug
Hi,
Thanks to Julian, my internal/external Nat problem is solved.
For anyone else working from outdated example files, the format with
localnet and localmask on separate lines is no longer supported.
The localnet line must also have the netmask included as per Julian's
example below or it will be ignored (or at least not work as expected).
Robert.
> -Original Message-
> From: Julian J. M.
> Sent: 23 July 2006 10:54
> Subject: Re: [asterisk-users] NAT and externip problem or bug
>
> Why don't you use the syntax that I mentioned in my first reply?
>
> According to
> http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+localnet
>
> The correct syntax is:
>
> localnet=192.168.0.0/255.255.255.0
>
> Keyword localmask is deprecated in asterisk 1.2... And btw,
> you should have seen it in the logs. According to chan_sip.c,
> around line 12508:
>
> } else if (!strcasecmp(v->name, "localmask")) {
> ast_log(LOG_WARNING, "Use of
> localmask is no long supported -- use localnet with mask syntax\n");
> }
>
>
> Julian J. M.
>
> On 7/22/06, Robert Jenkins <[EMAIL PROTECTED]> wrote:
> > The simple thing is that if I have 'externip' set, I can
> see on a soft
> > phone (running on a PC on the same local subnet as
> asterisk) that it's
> > seeing a call from another local device as coming from
> > [EMAIL PROTECTED] - which is the external IP and as everything is
> > inside the firewall there is no audio from the soft phone
> when the call answered.
> >
> > If I comment out the 'externip' line & restart asterisk, the soft
> > phone then correctly sees the local call as being from
> > [EMAIL PROTECTED] and I get two-way speech.
>
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Re: [asterisk-users] termcap support not found
- [EMAIL PROTECTED] wrote: > Im trying to install asterisk 1.2.10 on a new debian 3.1r2 machine > and every > time i try to make it i get an > > Configure: error: termcap support not found > Make: *** [editline/libedit.a] Error 1 Install the libncurses-dev package. -- Russell Bryant Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dial Plan to Play Message
which is exactly what I said if you read the whole thread :-) Eric "ManxPower" Wieling wrote: You can do it one of two ways: 1) make the SIP device dial a predefined number when the user picks up the phone. You do this in the SIP device. Check the manual for that device for detail on how to do this. It's normally called "hotline". In extensions.conf have Asterisk run Authenticate before the Dial() line. 2) Let the SIP device dial as normal, but in the dialplan execute Authenticate before the Dial line. Steve Totaro wrote: You could put the phone in a context such as context=restricted in sip.conf In extensions.conf put a context [restricted] exten => _.,1,Answer exten => _.,2,Authenticate(8675301) exten => _.,3,Goto(whateverdialcontext,whateverexten,whateverpriority) replace Allison's recording for authenticate with your own. Unless I am totally missing what you are trying to do. Thanks, Steve Eric "ManxPower" Wieling wrote: "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. Do you really want them to be different? Dial() will stop processing of the dialplan until the call ends. Do you really want this? "r" option to Dial will force a ringing sound to the caller, even if the caller should be hearing a "all circuits are busy", or "your call cannot be completed as dialed" or similar message. Do you really want that? [EMAIL PROTECTED] wrote: Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels. exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transfer exten => 8407,n,Authenticate(9461) exten => 8407,n,Playback(pbx-invalid) exten => 8407,n,Hangup() and in sip.conf [9507] type=friend user=8407 secret=xx ;context=from-sip callerid=8407 host=dynamic nat=yes qualify=yes canreinvite=no dtmfmode=rfc2833 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dial Plan to Play Message
You can do it one of two ways: 1) make the SIP device dial a predefined number when the user picks up the phone. You do this in the SIP device. Check the manual for that device for detail on how to do this. It's normally called "hotline". In extensions.conf have Asterisk run Authenticate before the Dial() line. 2) Let the SIP device dial as normal, but in the dialplan execute Authenticate before the Dial line. Steve Totaro wrote: You could put the phone in a context such as context=restricted in sip.conf In extensions.conf put a context [restricted] exten => _.,1,Answer exten => _.,2,Authenticate(8675301) exten => _.,3,Goto(whateverdialcontext,whateverexten,whateverpriority) replace Allison's recording for authenticate with your own. Unless I am totally missing what you are trying to do. Thanks, Steve Eric "ManxPower" Wieling wrote: "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. Do you really want them to be different? Dial() will stop processing of the dialplan until the call ends. Do you really want this? "r" option to Dial will force a ringing sound to the caller, even if the caller should be hearing a "all circuits are busy", or "your call cannot be completed as dialed" or similar message. Do you really want that? [EMAIL PROTECTED] wrote: Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels. exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transfer exten => 8407,n,Authenticate(9461) exten => 8407,n,Playback(pbx-invalid) exten => 8407,n,Hangup() and in sip.conf [9507] type=friend user=8407 secret=xx ;context=from-sip callerid=8407 host=dynamic nat=yes qualify=yes canreinvite=no dtmfmode=rfc2833 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P clone not working
Tom Lynn: > perhaps not what you're looking for, but reading thru your config, it looks > like you've mis-spelled 'echo cancel' as 'echo cancle' you are right, typo thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dial Plan to Play Message
You could put the phone in a context such as context=restricted in sip.conf In extensions.conf put a context [restricted] exten => _.,1,Answer exten => _.,2,Authenticate(8675301) exten => _.,3,Goto(whateverdialcontext,whateverexten,whateverpriority) replace Allison's recording for authenticate with your own. Unless I am totally missing what you are trying to do. Thanks, Steve Eric "ManxPower" Wieling wrote: "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. Do you really want them to be different? Dial() will stop processing of the dialplan until the call ends. Do you really want this? "r" option to Dial will force a ringing sound to the caller, even if the caller should be hearing a "all circuits are busy", or "your call cannot be completed as dialed" or similar message. Do you really want that? [EMAIL PROTECTED] wrote: Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels. exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transfer exten => 8407,n,Authenticate(9461) exten => 8407,n,Playback(pbx-invalid) exten => 8407,n,Hangup() and in sip.conf [9507] type=friend user=8407 secret=xx ;context=from-sip callerid=8407 host=dynamic nat=yes qualify=yes canreinvite=no dtmfmode=rfc2833 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P clone not working
perhaps not what you're looking for, but reading thru your config, it looks like you've mis-spelled 'echo cancel' as 'echo cancle'On 7/23/06, Frank Darner <[EMAIL PROTECTED]> wrote: > > What is the output from 'cat /proc/zaptel/*'>> After delete of all Asterisk files and complete new install I got> something:>> # modprobe zaptel && modprobe wcfxo >> linux:/proc/zaptel # cat /proc/zaptel/*> Span 1: WCFXO/0 "Generic Clone Board 1" RED>>1 WCFXO/0/0>> but # ztcfg -> is still no channels. >> any ideas?>OK, I have now an output.For some reason ztcfg was not looking in /etc/asterisk for the zaptel.conf.After using the option -c /etc/asterisk/zaptel.conf everythink was fine. One last thing:Playback sounds now like MickeyMouse, much to slowalso Calls SIP to SIP are not any more possibleany ideas?___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Woes
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello all, I've been trying to play with asterisk (after a two month break) and am having some problems getting my SIP connection to a third party provider to work. In the asterisk console I notice: - - debian*CLI> set verbose 999 Verbosity was 0 and is now 999 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:2355 sip_alloc: Allocating new SIP call for [EMAIL PROTECTED] Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:5441 check_user_full: Setting NAT on RTP to 4 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:840 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:5441 check_user_full: Setting NAT on RTP to 4 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:7329 handle_request: Check for res for 200 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:1620 update_user_counter: Call from user '200' is 1 out of 0 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:840 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found - - I believe that's some sort of SIP routing issue related to ReInvite's ? - - Is there a workaround for this? In the attempt that someone may be able to shed some light on the matter, I've uploaded my current configuration to: http://files.davehope.co.uk/asterisk-problem/ I've also uploaded the output of 'sip debug'. The interesting bit in that (to me at least) is the message: - - Looking for 10 in Outgoing Reliably Transmitting (NAT): SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP - - Is it so simple that I've missed something out in my outgoing bit on my dialplan ? Anyway, the complete log can be found here: http://files.davehope.co.uk/asterisk-problem/debug.log Ohh. And: - - [EMAIL PROTECTED]:/etc/asterisk# asterisk -V Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k - - If anyone would be so kind as to shed some insight into the matter it'd be greatly appreciated!, Kind Regards, Dave -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEw4ywjdL3ZT1KDlERAvgqAJ9ptCZlpKeFDkdKNaOHBKDLHi3HrgCglG3I 5K48wq9FfL4VlBkADOtLvXU= =57su -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spoofing a BLF Signal?
On Wed, May 24, 2006 at 08:54:09PM -0400, Matt wrote: > Then bristuff may be the way to go. However, I read this on the wiki > "Note: Using bristuff breaks PRI support, so you cant have both bri > and pri in the same server. " > That's not good! I need PRI. I never had this problem, using a BRI and a PRI on the same machine with bristuff patches. Ok, a quite old version of them, though. The only side effet of bristuff is that it will prevent you to link proprietary (non GPL) driver into it and redistribute. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dial Plan to Play Message
That was a typo its corrected to [8407] but problem still persist with original questions though. -- Original message -- From: "Eric "ManxPower" Wieling" <[EMAIL PROTECTED]> > "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. > Do you really want them to be different? > > Dial() will stop processing of the dialplan until the call ends. Do you > really want this? > > "r" option to Dial will force a ringing sound to the caller, even if the > caller should be hearing a "all circuits are busy", or "your call cannot > be completed as dialed" or similar message. Do you really want that? > > [EMAIL PROTECTED] wrote: > > Thanks for the response, its looks logical, for some reason the authentication > is not working for me, I'm using a Linksys Phone adapter and here is a sample > dial plan in extensions.conf and also SIP channels. & gt; > > > exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transfer > > exten => 8407,n,Authenticate(9461) > > exten => 8407,n,Playback(pbx-invalid) > > exten => 8407,n,Hangup() > > > > and in sip.conf > > > > [9507] > > type=friend > > user=8407 > > secret=xx > > ;context=from-sip > > callerid=8407 > > host=dynamic > > nat=yes > > qualify=yes > > canreinvite=no > > dtmfmode=rfc2833 > > -- > Now accepting new clients in Birmingham, Atlanta, Huntsville, > Chattanooga, and Montgomery. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk -users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dial Plan to Play Message
"[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. Do you really want them to be different? Dial() will stop processing of the dialplan until the call ends. Do you really want this? "r" option to Dial will force a ringing sound to the caller, even if the caller should be hearing a "all circuits are busy", or "your call cannot be completed as dialed" or similar message. Do you really want that? [EMAIL PROTECTED] wrote: Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels. exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transfer exten => 8407,n,Authenticate(9461) exten => 8407,n,Playback(pbx-invalid) exten => 8407,n,Hangup() and in sip.conf [9507] type=friend user=8407 secret=xx ;context=from-sip callerid=8407 host=dynamic nat=yes qualify=yes canreinvite=no dtmfmode=rfc2833 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dial Plan to Play Message
Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels. exten => 8407,1,Dial(SIP/8407,80,rt) ; permit transferexten => 8407,n,Authenticate(9461) exten => 8407,n,Playback(pbx-invalid)exten => 8407,n,Hangup() and in sip.conf [9507]type=frienduser=8407secret=xx;context=from-sipcallerid=8407host=dynamicnat=yesqualify=yescanreinvite=nodtmfmode=rfc2833;incominglimit=1;[EMAIL PROTECTED];disallow=all;allow=ulaw;allow=alaw;allow=g729;allow=g723.1 I also tried changing type to peer instead of freind and does not work either. I am running Asterisk 1.2.3. Any help will be appreciated and thanks for all your inputs. -- Original message -- From: Steve Totaro <[EMAIL PROTECTED]> > If you are using phones attached to a ZAP FXS port the immediate=yes > will work. Otherwise, some SIP phones (Grandstream for instance) allows > you to enter an autodial number. It depends on what is providing the > dialtone to the handset. If your device does not support autodial, then > the next best thing is to do what has already been suggested. > > OR > > [somecontext] > exten=s,1,answer > exten=s,2,Authenticate(insertdigitshere) > exten=s,3,(continue with a real dialplan) > > Change the corresponding authenticate gsm file to say what you want > about contacting the boss. > > This gives the impression that phone is restricted for outbound calling > b ut if you enter the authenticate string, you can dialout for > emergencies or convenience. > > Thanks, > Steve > > [EMAIL PROTECTED] wrote: > > It did not work, how can I put in some user intervention so that any > > numbers they dial will send them to a message? Restrict their outbound > > calls and a get a message to contact administrator instead of a busy > > signal. > > > > -- Original message -- > > From: "brandon kruz" <[EMAIL PROTECTED]>> > > > > thank you russel > > > forgot to mention this. > > > > > > > > > > >On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote: > > > > > [internal] > > > > > exten => s,1,Answer() > > > > > exten => s,n,Playback(custom) > > > > > exten => s,n,Hangup() > > > > > > > >This, by itself, does not solve the problem where you want the > > message > > > >to be played back when the phone is picked up without any user > > > >intervention. If you're using zap phones, you can simply set this > > > >option: > > > > > > > >immediate=yes > > > > > > > >Then, as soon as the phone goes off hook, the call will begin > > at the 's' > > > >extension in the configured context instead of pro viding > > dialtone. > > > > > > > >-- > > > >Russell Bryant > > > >Software Developer > > > >Digium, Inc. > > > > > > > >___ > > > >--Bandwidth and Colocation provided by Easynews. com -- > > > > > > > >asterisk-users mailing list > > > >To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _ > > > Dont just search. Find. Check out the new MSN Search! > > > http://search.msn.click-url.com/go/onm00200636ave/direct/01/ > > > > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > __ _ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dial Plan to Play Message
If you are using phones attached to a ZAP FXS port the immediate=yes will work. Otherwise, some SIP phones (Grandstream for instance) allows you to enter an autodial number. It depends on what is providing the dialtone to the handset. If your device does not support autodial, then the next best thing is to do what has already been suggested. OR [somecontext] exten=s,1,answer exten=s,2,Authenticate(insertdigitshere) exten=s,3,(continue with a real dialplan) Change the corresponding authenticate gsm file to say what you want about contacting the boss. This gives the impression that phone is restricted for outbound calling but if you enter the authenticate string, you can dialout for emergencies or convenience. Thanks, Steve [EMAIL PROTECTED] wrote: It did not work, how can I put in some user intervention so that any numbers they dial will send them to a message? Restrict their outbound calls and a get a message to contact administrator instead of a busy signal. -- Original message -- From: "brandon kruz" <[EMAIL PROTECTED]> > thank you russel > forgot to mention this. > > > > >On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote: > > > [internal] > > > exten => s,1,Answer() > > > exten => s,n,Playback(custom) > > > exten => s,n,Hangup() > > > >This, by itself, does not solve the problem where you want the message > >to be played back when the phone is picked up without any user > >intervention. If you're using zap phones, you can simply set this > >option: > > > >immediate=yes > > > >Then, as soon as the phone goes off hook, the call will begin at the 's' > >extension in the configured context instead of pro viding dialtone. > > > >-- > >Russell Bryant > >Software Developer > >Digium, Inc. > > > >___ > >--Bandwidth and Colocation provided by Easynews.com -- > > > >asterisk-users mailing list > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _ > Don’t just search. Find. Check out the new MSN Search! > http://search.msn.click-url.com/go/onm00200636ave/direct/01/ > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G726 codec softphone
Anybody know about a softphone (open source) that support G726-32 codec ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline Looking for Linux Virtual Private Servers ? Click here: http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426&a_bid=56 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P clone not working
> > What is the output from 'cat /proc/zaptel/*' > > After delete of all Asterisk files and complete new install I got > something: > > # modprobe zaptel && modprobe wcfxo > > linux:/proc/zaptel # cat /proc/zaptel/* > Span 1: WCFXO/0 "Generic Clone Board 1" RED > >1 WCFXO/0/0 > > but # ztcfg - > is still no channels. > > any ideas? > OK, I have now an output. For some reason ztcfg was not looking in /etc/asterisk for the zaptel.conf. After using the option -c /etc/asterisk/zaptel.conf everythink was fine. One last thing: Playback sounds now like MickeyMouse, much to slow also Calls SIP to SIP are not any more possible any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Dial Plan to Play Message
It did not work, how can I put in some user intervention so that any numbers they dial will send them to a message? Restrict their outbound calls and a get a message to contact administrator instead of a busy signal. -- Original message -- From: "brandon kruz" <[EMAIL PROTECTED]> > thank you russel > forgot to mention this. > > > >From: Russell Bryant <[EMAIL PROTECTED]>> >Reply-To: Asterisk Users Mailing List - Non-Commercial > >Discussion > >To: Asterisk Users Mailing List - Non-Commercial > >Discussion > >Subject: RE: [asterisk-users] Asterisk Dial Plan to Play Message > >Date: Sat, 22 Jul 2006 21:29:23 -0400 > >MIME-Version: 1.0 > >Received: from lists.digium.com ([69.16.138.164]) by > >bay0-mc9-f18.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, > >22 Jul 2006 18:32:43 -0700 > >Received: from digium-69-16-138-164.phx1.puregig.net ( localh ost > >[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 563422FC409;Sat, 22 > >Jul 2006 18:29:35 -0700 (MST) > >Received: from abita.digium.internal (gateway.digium.com [216.207.245.1])by > >lists.digium.com (Postfix) with ESMTP id 9CD5C2FC25Cfor > >;Sat, 22 Jul 2006 18:29:24 -0700 (MST) > >Received: from heineken.digium.com (heineken.digium.internal [10.16.1.2])by > >abita.digium.internal (Postfix) with ESMTP id C536AA94939for > >;Sat, 22 Jul 2006 20:29:25 -0500 (CDT) > >Received: from [172.17.99.18] ([172.17.99.18])by heineken.digium.com > >(8.13.6/69.69.69) with ESMTP id k6N1UIr9030086for > >; Sat, 22 Jul 2006 20:30:19 -0500 > >X-Message-Info: LsUYwwHHNt14xbUYi+9bCaWgpoxRQZbXIFwSWMVl+QA= > >X-Original-To: asterisk-users@lists.digium.com > & gt;Delivered-To: asterisk-users@lists.digium.com > >References: <[EMAIL PROTECTED]>> >Organization: Digium, Inc. > >X-Mailer: Evolution 2.6.1 X-BeenThere: asterisk-users@lists.digium.com > >X-Mailman-Version: 2.1.5 > >Precedence: list > >List-Id: Asterisk Users Mailing List - Non-Commercial > >Discussion > >List-Unsubscribe: > >,> [EMAIL PROTECTED]> > >List-Archive: > >List-Post: > >List-Help: > >List-Subscribe: > >,> [EMAIL PROTECTED] igium. com?subject=subscribe> > >Errors-To: [EMAIL PROTECTED] > >Return-Path: [EMAIL PROTECTED] > >X-OriginalArrivalTime: 23 Jul 2006 01:32:44.0557 (UTC) > >FILETIME=[E58083D0:01C6ADF7] > > > >On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote: > > > [internal] > > > exten => s,1,Answer() > > > exten => s,n,Playback(custom) > > > exten => s,n,Hangup() > > > >This, by itself, does not solve the problem where you want the message > >to be played back when the phone is picked up without any user > >intervention. If you're using zap phones, you can simply set this > >option: > > > >immediate=yes > > > >Then, as soon as the phone goes off hook, the call will begin at the 's' > >extension in the configured context instead of pro viding dialtone. > > > >-- > >Russell Bryant > >Software Developer > >Digium, Inc. > > > >___ > >--Bandwidth and Colocation provided by Easynews.com -- > > > >asterisk-users mailing list > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _ > Dont just search. Find. Check out the new MSN Search! > http://search.msn.click-url.com/go/onm00200636ave/direct/01/ > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P clone not working
Am Sunday 23 July 2006 06:39 schrieb Tzafrir Cohen: > On Sat, Jul 22, 2006 at 09:02:48PM +0200, Frank Darner wrote: > > Hi, > > > > I have problem to set up an X100P clone card. > > Installation of zaptel was successful. > > Also modprobe of zaptel, ztdummy and wcfxo without problems. > > > > kernel: wcfxo: module not supported by Novell, setting U taint flag. > > kernel: ACPI: PCI Interrupt :05:04.0[A] -> GSI 16 (level, low) -> IRQ > > 193 kernel: wcfxo: DAA mode is 'FCC' > > kernel: Found a Wildcard FXO: Generic Clone > > kernel: Registered tone zone 0 (United States / North America) > > > > > > But if I do #ztcfg -vvv no channels will be found. > > What is the exact output from that command? > > What is the output from 'cat /proc/zaptel/*' After delete of all Asterisk files and complete new install I got something: # modprobe zaptel && modprobe wcfxo linux:/proc/zaptel # cat /proc/zaptel/* Span 1: WCFXO/0 "Generic Clone Board 1" RED 1 WCFXO/0/0 but # ztcfg - is still no channels. any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] termcap support not found
This is probably an easy one but i have not been able to fix it. Im trying to install asterisk 1.2.10 on a new debian 3.1r2 machine and every time i try to make it i get an Configure: error: termcap support not found Make: *** [editline/libedit.a] Error 1 Ive installed termcap-compat using apt-get but this has not solved the problem. Can anybody tell me what im doing/not doing properly. From matt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT and externip problem or bug
Why don't you use the syntax that I mentioned in my first reply?
According to http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+localnet
The correct syntax is:
localnet=192.168.0.0/255.255.255.0
Keyword localmask is deprecated in asterisk 1.2... And btw, you should
have seen it in the logs. According to chan_sip.c, around line 12508:
} else if (!strcasecmp(v->name, "localmask")) {
ast_log(LOG_WARNING, "Use of localmask is no
long supported -- use localnet with mask syntax\n");
}
Julian J. M.
On 7/22/06, Robert Jenkins <[EMAIL PROTECTED]> wrote:
The simple thing is that if I have 'externip' set, I can see on a soft phone
(running on a PC on the same local subnet as asterisk) that it's seeing a
call from another local device as coming from [EMAIL PROTECTED] - which is
the external IP and as everything is inside the firewall there is no audio
from the soft phone when the call answered.
If I comment out the 'externip' line & restart asterisk, the soft phone then
correctly sees the local call as being from [EMAIL PROTECTED] and I get
two-way speech.
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Re: [asterisk-users] question about asterisk DB
Unplug, I'm sure there are other people with better ideas but if you see on sineapps, I remember someone having written a patch which seperates out the the sip registry into a new table. If this is stable and tested, then you might want to use that with an ARA configuration and have all your asterisk servers be able to lookup registry info of UAs regardless of which (*) server they've come in on. I dunno if this works properly, but either this or sticking in SER in front of your (*) servers might be a better idea than fighting your way with the astdb (with all due respect). On 7/23/06, unplug <[EMAIL PROTECTED]> wrote: Thanks! Actually, I want to share the asterisk DB using multiple asterisks. So I use NFS to share the whole directory /var/lib/asterisk in order to share files including astdb of asterisk. However, there is not what I expected. Say, UA1 registers asterisk1 and UA2 registers asterisk2. Asterisk1 can only know UA1 and asterisk2 can only know UA2. It is what I find in CLI with database show. So I want to know more about astdb (where it store data?) in order to share it between multiple asterisk. If /var/lib/asterisk/astdb is used to store data of asterisk DB, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P clone not working
> configs look fine to me > it has to be a module problem im guessing may be, but I dont know whats wrong Zaptel (1.2.6.) installation was without errors and I did not get an error when loading the modules > > try rmmod for zaptel and wctfxo or w/e then modprobing again(must be root) > > if that does not work(which is does on my system, so i added it into a > startup conf) > > >kernel: wcfxo: module not supported by Novell > > are you sure that wcfxo is the correct module for that device?? Yes, I doublechecked with other sites ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P clone not working
Tzafrir Cohen: > On Sat, Jul 22, 2006 at 09:02:48PM +0200, Frank Darner wrote: > > Hi, > > > > I have problem to set up an X100P clone card. > > Installation of zaptel was successful. > > Also modprobe of zaptel, ztdummy and wcfxo without problems. > > > > kernel: wcfxo: module not supported by Novell, setting U taint flag. > > kernel: ACPI: PCI Interrupt :05:04.0[A] -> GSI 16 (level, low) -> IRQ > > 193 kernel: wcfxo: DAA mode is 'FCC' > > kernel: Found a Wildcard FXO: Generic Clone > > kernel: Registered tone zone 0 (United States / North America) > > > > > > But if I do #ztcfg -vvv no channels will be found. > Hi, OK, I do # modprobe zaptel && modprobe wcfxo > What is the exact output from that command? linux:/proc/zaptel # ztcfg -vvv Zaptel Configuration == Channel map: 0 channels configured. > > What is the output from 'cat /proc/zaptel/*' nothing dir /proc/zaptel but is empty BUT, if I do additional # modprobe ztdummy linux:/proc/zaptel # cat /proc/zaptel/* Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" any ideas ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SOLVED:RE : [asterisk-users] asterisk-1.2.9 / chan-oh323.so
--- [EMAIL PROTECTED] a écrit :
> Hello,
>
> Asterisk crash with chan_oh323.so
> i use asterisk 1.2.9 asterisk-oh323-0.7.3
>
> What's wrong ?
>
>
ACF|192.168.0.11:1720|4762_endp|2505|900:dialedDigits|903:h323_ID=903:dialedDigits=903:h323_ID=903:dialedDigits|false;
>
ACF|80.119.15.247:1721|4760_endp|2505|asterisk-gw:h323_ID|harry
> gaillac [80.119.15.247]:h323_ID|true;
> /usr/sbin/safe_asterisk: line 84: 6206 Segmentation
> fault asterisk ${CLIARGS} ${ASTARGS}
> 1>&/dev/${TTY} Asterisk ended with exit status 139
> Asterisk exited on signal 11.
> Automatically restarting Asterisk.
> DCF|192.168.0.11|4762_endp|2505|normalDrop;
> GCF|80.119.15.247|asterisk-gw:h323_ID|gateway;
>
RCF|80.119.15.247:1721|asterisk-gw:h323_ID|gateway|4760_endp;
> IRQ|192.168.0.11:1042|4762_endp;
> GCF|213.115.171.140|901:dialedDigits|gateway;
> GCF|213.115.171.140|901:dialedDigits|gateway;
>
RCF|192.168.178.14:1720|901:dialedDigits|gateway|4763_endp;
>
>
>
>
>
>
>
>
>
>
>
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[asterisk-users] Request for some help....
Hi,
I am working out to establish calls
between two asterisks PBX'es using H323 channels. I am using SJ phones as
the H323 clients.
The scenario looks like,
SJ phone(444) --- Asterisk PBX 1 --- H323 channel ---
Asterisk PBX2 --- SJ phone(555).
I was able to make the calls and channel was successfully established. At
both the SJ Phones it displays call is operational.
But the voice is not getting transmitted, I was not able to hear
anything at both the ends. Two asterisks are on machines 10.238.115.223 &
10.238.115.224, two clients are on the machines 10.238.115.226 and
10.238.115.230. I am including the H323.conf and extensions.conf of both the
asterisks.
Can someone kindly help me out to carry my work further.
H323.conf (Asterisk 10.238.115.223)
[general]
port = 1720bindaddr = 0.0.0.0allow=alawallow=ulaw
gatekeeper = DISABLE AllowGKRouted = yes
context=default
;alias definition
[det-gw/10.238.115.223]type=h323prefix=5,4allow=alawallow=ulawcontext=default [444]type=friendhost=10.238.115.226allow=alawallow=ulaw
extensions.conf (Asterisk 10.238.115.223)
[default]
exten => 444,1,Dial,H323/${EXTEN}
exten => 555,1,Dial,H323/[EMAIL PROTECTED]
H323.conf (Asterisk 10.238.5.224)
[general]
port = 1720bindaddr = 0.0.0.0allow=alawallow=ulaw
gatekeeper = DISABLE AllowGKRouted = yes
context=default
;alias definition
[det-gw/10.238.115.224]type=h323prefix=5,4allow=alawallow=ulawcontext=default [555]type=friendhost=10.238.115.230allow=alawallow=ulaw
extensions.conf (Asterisk 10.238.5.224)
exten => 555,1,Dial,H323/${EXTEN}
exten => 444,1,Dial,H323/[EMAIL PROTECTED]
Thanks in advance..
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Re: [asterisk-users] Operator Console(s)/Shared Call Appearances
Hi Mr. Jones, Mr. Jones wrote : It seems there are probably two routes, but I'm not sure of the limitations of each. 1. Shared call appearances. This would seem to be the most similar to what we currently have where we have stations/DNs for 3 executives on 3 assistants phones. Of course with the existing system we have lots of programmable buttons. We're leaning towards the SPA-942s, so I'd be interested to know how this might work (do we need the 4 line license, and are we limited to 4 call appearances)? SPA-942's look nice, I however have no experience in using them, but I don't really understand what you're trying to do this way. Do you want some kind of audio or other message so the receptionist knows when he or she can intercept a call? 2. Some form of PC application such as the one at Asternic.org, or something else. This would seem to have the most flexibility, but may require the operator to pay too much attention to the window, unless there's some audible notification. Still not what I should do in this situation... A couple of other alternatives maybe to create a queue, or possibly go with a "side car" type device. Well, it seems you haven't got very much experience in deploying * systems, which is absolutely no problem. What do you think about this idea; 1. Call comes in at one of the executive numbers. 2. Executive phone starts ringing for a predetermined time. 3. The callerid is changed to also reflect the name/number of called executive, so that the receptionist knows for who the call was. 4. The call is dropped into a queue for the receptionist (queue because multiple calls to the receptionist at the same time are possible). This setup isn't all that hard, and doesn't require more than 4 sip accounts / phones and one queue, with one agent. Furthermore, if your company starts to grow, and more receptionists that have to answer the phone are needed, it's quite easy, all you have to do is add a sip account, one agent and add that agent to the existing queue. (About 2 minutes...) -- Sebastian Berm iPronto Communications ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Load balenced (ADSL) network connections, is it possible?
> > > I need to put an Asterisk server in a remote office where only ADSL is > > available. Maximum of 8meg downstream 646k upstream. > > Is this an adsl2 line ? If yes ask your provider if it supports channel bonding. You could use 2 adsl lines as one. All "load balancing" etc is done at the dslam side. Also if annex M is supported you can get up to 3,5 mbits of upload (theoreticaly, usually is close to 2mbits, heavily depends on distance and line conditions) If you need to load balance at your side (ie the office) it can be done but would require setting up 2 * servers connected to each other and using some form of round robing the dns so that requests reach both servers. If the dsl line is also used for other purposes than voip, make sure that you use qos or you will be facing problems with the quality of the calls. Stelios ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
