Re: [asterisk-users] How to connect two asterisk server
Hi I would suggest a IAX2 trunk between the two servers. You will need to modify the dialplan to recognise which extensions are on each box and route accordingly. The fact your clients are SIP does not preclude you from using IAX2 to connect the servers. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Thirumal Saminathan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Thursday, 28 December, 2006 6:09:25 AM Subject: [asterisk-users] How to connect two asterisk server Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients.. plz anyone suggest me Regards, Thiru ___ --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 ___ All New Yahoo! Mail – Tired of [EMAIL PROTECTED]@! come-ons? Let our SpamGuard protect you. http://uk.docs.yahoo.com/nowyoucan.html___ --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] res_perl with asterisk 1.4 compile problem
Dear all, now we have the same problem of res_perl compilation with asterisk 1.4. It is the same problem that was present when asterisk was upgraded to version 1.2. I hope Anthony Minessale will be able to solve that problem as he did on that case. But if any of you know a hack to this problem please let us know. Here is the same compile problem again: gcc -Wall -DRES_PERL_BASE=\/usr/local/res_perl\ -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -Wdeclaration-after-statement -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm -I/usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE -I/usr/src/asterisk -I/usr/src/asterisk/include -I. -c res_perl.c In file included from ./res_perl.h:17, from res_perl.c:17: /usr/src/asterisk/include/asterisk/module.h:204: warning: struct ast_channel declared inside parameter list /usr/src/asterisk/include/asterisk/module.h:204: warning: its scope is only this definition or declaration, which is probably not what you want In file included from ./res_perl.h:22, from res_perl.c:17: /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2424: error: syntax error before perl_mutex /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2424: warning: type defaults to `int' in declaration of `perl_mutex' /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2424: warning: data definition has no type or storage class /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2425: error: syntax error before perl_cond /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2425: warning: type defaults to `int' in declaration of `perl_cond' /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:2425: warning: data definition has no type or storage class In file included from /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perl.h:3988, from ./res_perl.h:22, from res_perl.c:17: /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:48: error: syntax error before PL_op_mutex /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:48: warning: type defaults to `int' in declaration of `PL_op_mutex' /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:48: warning: data definition has no type or storage class /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:52: error: syntax error before PL_dollarzero_mutex /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:52: warning: type defaults to `int' in declaration of `PL_dollarzero_mutex' /usr/local/lib/perl5/5.8.8/i686-linux-thread-multi/CORE/perlvars.h:52: warning: data definition has no type or storage class res_perl.c:492: warning: initialization from incompatible pointer type res_perl.c: In function `_update_perl': res_perl.c:837: warning: ISO C90 forbids mixed declarations and code res_perl.c: At top level: res_perl.c:921: warning: initialization from incompatible pointer type res_perl.c:1023: warning: initialization from incompatible pointer type res_perl.c: In function `_load_module': res_perl.c:1036: warning: ISO C90 forbids mixed declarations and code make: *** res_perl.o Error 1 Rg, Gentian ___ --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] 1.4.0, IMAP and Dovecot
On Thu, Dec 28, 2006 at 12:35:57PM +1300, Ray Jackson wrote:
Dan,
I have IMAP support working now with Courier IMAP. Since Courier (and
probably Dovecot) do not support a single authuser connection that may
access any mailbox, you have to omit the 'authuser' and 'authpassword'
settings in voicemail.conf and then add the username/password login per
extension... e.g.
Are you sure that this is an explicit support in the mail server?
Here's what Mark Crispin (the author of both UW-imapd and c-client)
wrote recently:
| Does UW-IMAP have an admin user? If so, where is it configured?
|
| It's hidden in the release notes file.
|
| Any user who is in a UNIX group called mailadm has administrator
| rights in UW imapd and ipop3d. Administrator rights are the right to
| log in as any other user.
|
| For c-client based client programs (mailutil, Pine, Alpine, etc.), the
| /authuser flag is used by the mail administrator. For example the
| mailbox name specifier:
|
| {imap.example.com/authuser=fred/user=joe}INBOX
|
| will open a connection to imap.example.com and log in as user
| joe using user fred's password, and then open joe's INBOX. This assumes
| that user fred is in group mailadm on the imap.example.com.
So can you do this trick manually? authenticate as one user and read
another user's mailbox?
Here's an example with root and pre-authentication. I figure that some
tricks with pam and such will get you further:
[EMAIL PROTECTED] MAIL=maildir:/home/tzafrir/Maildir /usr/lib/dovecot/imap
* PREAUTH [CAPABILITY IMAP4rev1 SORT THREAD=REFERENCES MULTIAPPEND
* UNSELECT LITERAL+ IDLE CHILDREN LISTEXT LIST-SUBSCRIBED NAMESPACE]
* Logged in as root
1 list *
* LIST (\HasNoChildren) . INBOX
1 OK List completed.
2 select INBOX
* FLAGS (\Answered \Flagged \Deleted \Seen \Draft)
* OK [PERMANENTFLAGS (\Answered \Flagged \Deleted \Seen \Draft \*)]
* Flags permitted.
* 0 EXISTS
* 0 RECENT
* OK [UIDVALIDITY 1161851409] UIDs valid
* OK [UIDNEXT 1] Predicted next UID
2 OK [READ-WRITE] Select completed.
* logout
* BYE Logging out
* OK Logout completed.
This is dovecot 0.99.14 on Debian Sarge. Note that I actually don't use
that imap mailbox normally.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Re: Asterisk Queues
Hi, Well, in our case, it seems that the issue was being caused by announcements. That is, someone in QUEUE1 would be waiting 15 minutes.. and QUEUE2 would be waiting 5 minutes. The person in QUEUE1 would be listening to 'we're sorry you are holding so long, if you'd like to leave a message, press 1 now.. otherwise continue to hold'... and while they were listening to that an agent would become available. Well.. the only queue available now is QUEUE2, so that caller would get thrown to an agent. Really what should happen is, even if Asterisk is playing an announcement in queue.. it should still consider the call 'active-in-the-queue' and yank it out of the announcement if an agent becomes available. On 12/27/06, Phil Hopkins [EMAIL PROTECTED] wrote: You posted a question on the asterisk-users forum in Aug. regarding the order that Asterisk would answer calls with multiple queues (not answering the call with the longest wait time regardless of the queue it is in). We have run into the same problem, which I consider to be a show stopper until we find a solution. As I understand the behavior of Asterisk that is the way it works. Seems wrong but ... I am working on a solution using realtime queues but it really isn't optimal. Did you ever find a solution? Phil Hopkins MIS Manager BexarMet Water District 2047 W. Malone San Antonio, TX 78225 office - 210-357-5753 cell - 210-279-9720 ___ --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] The Good, Bad and Scam VoIP Providers
Of course everyone is allowed to use VoIP... Asterisk is open! I think Dovid's point was more that this guy's website says he buys and sells precious metals and other random items, his postings on this list indicate that he installs PBXes and resells VoIP services, and then his private e-mails say that he's a PI. The PI thing sounds just like him trying to get those who attacked him to back off. Alex On 12/28/06, Kevin Walsh [EMAIL PROTECTED] wrote: Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --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] Checking voicemail from outside
Hi all, I'm sure this is a stupid question, but is there a way to check your voicemail by calling your extension from the outside? When I call my own extension from outside and hit pound or star, it just stops my greeting and gives me the beep. I'd like to call my extension and press a key and be prompted for my password. Otherwise the only way I can think to get around this is to create an extension that goes to voicemailmain(). Thanks in advance, Phil ___ --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] 1.4 and unicall
No update on unicall and 1.4? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Anton Krall |Sent: Tuesday, December 26, 2006 6:15 AM |To: asterisk-users@lists.digium.com |Subject: [asterisk-users] 1.4 and unicall | |Guys, anybody knows if 1.4 has support for unicall or if/which version of |unicall will compile on 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 ___ --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 two asterisk server
Thiru, You can connect them by SIP or IAX, it depends on what kind of media you will need to transfer. For audio only IAX is ok, if you 're going to use video, SIP is the option. Carlos Alperin _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thirumal Saminathan Sent: Thursday, December 28, 2006 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [asterisk-users] How to connect two asterisk server Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients.. plz anyone suggest me Regards, Thiru ___ --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] Checking voicemail from outside
Phil,
Add this to your extensions (I have mine in a macro)
exten = a,1,VoicemailMain(${ARG1}); If they press *, send to
Voicemail
so it should look like...
exten = s,1,Dial(${ARG2},13,${ARG3})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = a,1,VoicemailMain(${ARG1})
I have a few other things in there as well, but those are the lines that
should do what you want. When you press *, you are prompted for a password.
Rob
Phil Finkler wrote:
Hi all,
I’m sure this is a stupid question, but is there a way to check your
voicemail by calling your extension from the outside? When I call my
own extension from outside and hit pound or star, it just stops my
greeting and gives me the “beep”. I’d like to call my extension and
press a key and be prompted for my password. Otherwise the only way I
can think to get around this is to create an extension that goes to
voicemailmain().
Thanks in advance,
Phil
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Re: [asterisk-users] Re: Asterisk Queues
On Thursday 28 December 2006 13:33, Matt wrote: Hi, Well, in our case, it seems that the issue was being caused by announcements. That is, someone in QUEUE1 would be waiting 15 minutes.. and QUEUE2 would be waiting 5 minutes. Yep we noticed this too - it's a rather unfortunate side-effect; we found a very agreeable workaround, though :) Instead of making use of the announcement feature, we made a .WAV file which is 60 seconds long. The first 10 seconds are 'Sorry you have been waiting' and the remaining 50 seconds are the sound of telephone ringing... This way the customer gets the announcement and indication they are on hold, and Asterisk processes the queue more elegantly :) The only side-effect is sometimes a queue member will join in the middle of the announcement, but this is of little concern to us. Cheers, Gavin ___ --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] 1.4 and unicall
I asked the same a while ago, without any kind of conclusive answer. But you have to consider that these are special dates I just spent all night studying/modifying mfcr2.c to my needs but I've never looked at the unicall code or the asterisk channel API. With respect to MFC/R2, and according to what it saw, it seems fairly complete on the incoming part of the protocol, but the outgoing logic is kind of crude. I wonder if Steve Underwood is still actively working on it. BarZ Anton Krall wrote: No update on unicall and 1.4? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Anton Krall |Sent: Tuesday, December 26, 2006 6:15 AM |To: asterisk-users@lists.digium.com |Subject: [asterisk-users] 1.4 and unicall | |Guys, anybody knows if 1.4 has support for unicall or if/which version of |unicall will compile on 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 ___ --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
FW: [asterisk-users] cdr_addon_mysql.so did not register itself duringload
So no one else is having issues with MySQL and 1.4? I'm the only one? -Original Message- From: Savoy, Kevin - Williston, ND Sent: Wednesday, December 27, 2006 2:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] cdr_addon_mysql.so did not register itself duringload Well the addons from 1.4 are installed. This original Asterisk 1.2.x box was created by my predecessor and he had the cdr_addon_mysql.so and res_config_mysql.so files on a server that we copied to any new installation. I'm not sure where he got these files. As far as I can tell shouldn't the install of the addons create these files? If not where do I get them from? I've done a search on the server and those files do NOT exist. Otherwise can you tell me how to load the MySQL in Asterisk 1.4 to make it work? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Tuesday, December 26, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cdr_addon_mysql.so did not register itself duringload Savoy, Kevin - Williston, ND wrote: I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4 as well. I can place calls but I noticed the MySQL was writing out to the database. When doing an Asterisk load with asterisk - I saw the following: [Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module: Module 'cdr_addon_mysql.so' did not register its [Dec 26 11:02:08] WARNING[10029]: loader.c:607 load_resource: Module 'cdr_addon_mysql.so' could not be loaded. The module that is being loaded is not a 1.4 module. It is using the old way of module loading. You should make sure that you are using 1.4 addons and that they are installed. -- Joshua Colp 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 ___ --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] Background switch to different context
I am using the Background() function to ask for the extension, but the extensions are in a different context. Is there a way to tell Background() to look for the entered extensions in another context other than the currently running one? Thanks. Keith ___ --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] Background switch to different context
I am using the Background() function to ask for the extension, but the extensions are in a different context. Is there a way to tell Background() to look for the entered extensions in another context other than the currently running one? in that context you can do include = other-context hth ___ --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] Background switch to different context
If you type show application background on the *CLI, you can see all the options listed there. The last optional argument is the context that you want to use to look for extensions. On 12/28/06, Time Bandit [EMAIL PROTECTED] wrote: I am using the Background() function to ask for the extension, but the extensions are in a different context. Is there a way to tell Background() to look for the entered extensions in another context other than the currently running one? in that context you can do include = other-context hth ___ --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
FW: [asterisk-users] cdr_addon_mysql.so did not register itself duringload
Ok so I'm the only one not getting this to work. Maybe I'm doing something wrong. Here is the installation I'm using. Install Fedora Core 4 and do all the updates through yum. Then I install zdlib-devel, openssl-devel, newt-devel, gcc, gcc-c++ and then mysql and perl-DBD-MySQL all using yum install. Am I missing something? Something I'm installing I shouldn't be? After doing the Asterisk-Addons with ./configure, make and then make install as it instructs, the two files below do NOT exist anywhere on my system. Can I compile these manually? If so how? Help? Thanks -Original Message- From: Savoy, Kevin - Williston, ND Sent: Thursday, December 28, 2006 9:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: FW: [asterisk-users] cdr_addon_mysql.so did not register itself duringload So no one else is having issues with MySQL and 1.4? I'm the only one? -Original Message- From: Savoy, Kevin - Williston, ND Sent: Wednesday, December 27, 2006 2:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] cdr_addon_mysql.so did not register itself duringload Well the addons from 1.4 are installed. This original Asterisk 1.2.x box was created by my predecessor and he had the cdr_addon_mysql.so and res_config_mysql.so files on a server that we copied to any new installation. I'm not sure where he got these files. As far as I can tell shouldn't the install of the addons create these files? If not where do I get them from? I've done a search on the server and those files do NOT exist. Otherwise can you tell me how to load the MySQL in Asterisk 1.4 to make it work? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Tuesday, December 26, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cdr_addon_mysql.so did not register itself duringload Savoy, Kevin - Williston, ND wrote: I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4 as well. I can place calls but I noticed the MySQL was writing out to the database. When doing an Asterisk load with asterisk - I saw the following: [Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module: Module 'cdr_addon_mysql.so' did not register its [Dec 26 11:02:08] WARNING[10029]: loader.c:607 load_resource: Module 'cdr_addon_mysql.so' could not be loaded. The module that is being loaded is not a 1.4 module. It is using the old way of module loading. You should make sure that you are using 1.4 addons and that they are installed. -- Joshua Colp 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 ___ --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] 1.4 and unicall
I hope so, he is the only guy working on mfcr2 right now. I have unicall working on 1.2 perfectly but if there will be no unicall support for 1.4, that would be a show stopper unless we use a mfcr2 converter... anybody knows any? Something that can convert mfcr2 to pri? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Barzilai Spinak |Sent: Thursday, December 28, 2006 8:26 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] 1.4 and unicall | |I asked the same a while ago, without any kind of conclusive answer. |But you have to consider that these are special dates |I just spent all night studying/modifying mfcr2.c to my needs but |I've never looked at the unicall code or the asterisk channel API. |With respect to MFC/R2, and according to what it saw, it seems fairly |complete on the incoming part of the protocol, but the outgoing logic is |kind of crude. |I wonder if Steve Underwood is still actively working on it. | |BarZ | |Anton Krall wrote: | No update on unicall and 1.4? | | |-Original Message- | |From: [EMAIL PROTECTED] [mailto:asterisk-users- | |[EMAIL PROTECTED] On Behalf Of Anton Krall | |Sent: Tuesday, December 26, 2006 6:15 AM | |To: asterisk-users@lists.digium.com | |Subject: [asterisk-users] 1.4 and unicall | | | |Guys, anybody knows if 1.4 has support for unicall or if/which version of | |unicall will compile on 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 | | | | ___ | --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 BEGIN:VCARD VERSION:2.1 X-MS-SIGNATURE:YES N;LANGUAGE=en-us:Krall;Anton FN:Anton Krall ORG:Intruder Consulting TITLE:A Division of IntruderEnterprises S.A. de C.V. TEL;WORK;VOICE:+52 (55) 5781-5112 x 201 TEL;WORK;VOICE:+52 (55) 5985-2430 x 201 X-MS-OL-DEFAULT-POSTAL-ADDRESS:0 URL;WORK:http://www.intruder.com.mx EMAIL;PREF;INTERNET:[EMAIL PROTECTED] PHOTO;TYPE=JPEG;ENCODING=BASE64: /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa KCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCj/wAAR CAAXAEgDASIAAhEBAxEB/8QAHwAAAQUBAQEBAQEAAAECAwQFBgcICQoL/8QAtRAA AgEDAwIEAwUFBAQAAAF9AQIDAAQRBRIhMUEGE1FhByJxFDKBkaEII0KxwRVS0fAkM2JyggkK FhcYGRolJicoKSo0NTY3ODk6Q0RFRkdISUpTVFVWV1hZWmNkZWZnaGlqc3R1dnd4eXqDhIWG h4iJipKTlJWWl5iZmqKjpKWmp6ipqrKztLW2t7i5usLDxMXGx8jJytLT1NXW19jZ2uHi4+Tl 5ufo6erx8vP09fb3+Pn6/8QAHwEAAwEBAQEBAQEBAQECAwQFBgcICQoL/8QAtREA AgECBAQDBAcFBAQAAQJ3AAECAxEEBSExBhJBUQdhcRMiMoEIFEKRobHBCSMzUvAVYnLRChYk NOEl8RcYGRomJygpKjU2Nzg5OkNERUZHSElKU1RVVldYWVpjZGVmZ2hpanN0dXZ3eHl6goOE hYaHiImKkpOUlZaXmJmaoqOkpaanqKmqsrO0tba3uLm6wsPExcbHyMnK0tPU1dbX2Nna4uPk 5ebn6Onq8vP09fb3+Pn6/9oADAMBAAIRAxEAPwDI8HeCb3xFHJclhb2MY+adumfQDua62b4Q iBGluNWEcOPlZreQZJ/4DXS/DAXn/CBxS6fArXscc5gbPOflGfrgmun8NNfr4dt5vESs2oG2 lRNxy7ZZcYHryfwzXo4nGVouTi0knY+ew+CoyhHmV21e58/XvhaWz1XWLGZpC+n27TkxoGBw QOckYHPXn6VFdeDdftbV7i4050iRFkbMiblRsYYrnIXkc4xXWeJNRFp408Xosc9217YyW6mF d20llO5vQDFUdQ8TxT6pr2pi0vBaalpqafCzLwHEcanJzjGUbpXoxqVGk/66HnzpUU2n3f6/ 8D7zO1D4f67bao9lbQJdukEdw7xyKFRXAPJJ4wTjJ9M9Kxl0DVHu7a1Wzcz3MJuIUBHzxgMS w56YVj+FdnrevxXMesW8FjqaX2oaba27RPDjyzDsBPXJUhOuO9LpPijRoDo+p3UOp/bNP017 ARxxqYmyroH3k5/j6Y696FUqct2v6t/mEqNFysnZf8H07amJ4Z8CarrU1o0sRtbK4jeVZmZS 2xVJ3CMsGKkjGcY5rNs/CutXmmrfW1i727Kzod6hnVfvMqE7mA55ANdvYa7psOraXq9/b6ol /YW/9lNBEiGFpVRkBD7hjg5K4696bpnjOC30XSrqSC9huNNtjZqYrOFkkYbtpEzAsn3uQAf1 pOpVvov61/4BSo0LJN/1p/wdNzg9Q0DU9P0221C9tWhtLkBoXZl+cEZBAznGO+KKveM9Qa/b Rw1vcQfZtOhtsTLt3Fc5ZfbNFdEG2rs46qjGVo7Gh4Z8e3uhaetpHbW9wiZ8syru2Z64BrRn +KmqPaslva2ttcFCnnxIFYA9en86KKyeGpSlzOKuaRxdaK5VLQ4qHVLiOaeRisrTDD+YM8g5 B+oIpG1KdtPSzITy0xhsfMQCzAfTLMfxoorblRjzPuWItevY74XeY2l2svK8EM5c5APPzMeO nY8VX/tGb+zjZbY/LJ+9t+bGc4z6Zooo5UHPLuOOq3JnMx2eYbn7Xnb0fOfyqP8AtCf7IbbK +Ue2OfvBv5iiijlQcz7ljXNUGqSwuIBDsVt2GzuZmLM34k0UUU0klZClJyd2f//Z X-MS-OL-DESIGN;CHARSET=utf-8:card xmlns=http://schemas.microsoft.com/office/outlook/12/electronicbusinesscards; ver=1.0 layout=top bgcolor=ffimg xmlns= align=tright area=25 use=photo/fld xmlns= prop=name align=left dir=ltr style=b color=00 size=10/fld xmlns= prop=org align=left dir=ltr color=00 size=8/fld xmlns= prop=title align=left dir=ltr color=00 size=8/fld xmlns= prop=blank
Re: FW: [asterisk-users] cdr_addon_mysql.so did not register itself duringload
Savoy, Kevin - Williston, ND wrote: Ok so I'm the only one not getting this to work. Maybe I'm doing something wrong. Here is the installation I'm using. Install Fedora Core 4 and do all the updates through yum. Then I install zdlib-devel, openssl-devel, newt-devel, gcc, gcc-c++ and then mysql and perl-DBD-MySQL all using yum install. Am I missing something? Something I'm installing I shouldn't be? After doing the Asterisk-Addons with ./configure, make and then make install as it instructs, the two files below do NOT exist anywhere on my system. Can I compile these manually? If so how? Help? What does ./configure say for MySQL? Should be two lines: checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... yes If it's like the above then the CDR module should compile. -- Joshua Colp 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] cdr_addon_mysql.so did not register itselfduringload
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Savoy, Kevin - Williston, ND Sent: Thursday, December 28, 2006 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: FW: [asterisk-users] cdr_addon_mysql.so did not register itselfduringload Ok so I'm the only one not getting this to work. Maybe I'm doing something wrong. Here is the installation I'm using. Install Fedora Core 4 and do all the updates through yum. Then I install zdlib-devel, openssl-devel, newt-devel, gcc, gcc-c++ and then mysql and perl-DBD-MySQL all using yum install. Am I missing something? Something I'm installing I shouldn't be? After doing the Asterisk-Addons with ./configure, make and then make install as it instructs, the two files below do NOT exist anywhere on my system. Can I compile these manually? If so how? Help? You will also need mysql-devel, and if you pay close attention to the output of ./configure, it likely tells you that you don't have it. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy 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
[asterisk-users] Re: Voicemail hangup by gateway? Audiocodes
On 2006-12-24 00:35:06 -0800, Martin Joseph [EMAIL PROTECTED] said: I have a spiffy new gateway which seems quite promising. It's the Audiocodes MP114 FXS_FXO (2 of each). I have got it configured and working reasonably well, but have a couple of issues. 1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway after 10 seconds. This isn't asterisk saying it's quiet for 10 seconds, it's the gateway deciding it's time to go by by. Ok, I am still annoyed by this, I did an SIP debug and I can see that 10 seconds (after the beep) the gateways sends and SIP bye to Asterisk. I have exhaustively searched the Gateway's web interface looking for something that might be causing this, but can't find anything (enabled that is). Has anyone out there seen this with audiocodes hardware before? I assume it's due to the fact that when the voicemail system starts recording, there is no longer any audio going toward the gateway that it decides the call must be over. Hoping for some help. Marty ___ --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: FW: [asterisk-users] cdr_addon_mysql.so did not register itselfduringload
Ok so something is missing. I get the below for those two lines. checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... no I even installed the mysql-devel as Bradley Watkins suggested and still it says no. What do I need to make that say yes? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Thursday, December 28, 2006 11:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: FW: [asterisk-users] cdr_addon_mysql.so did not register itselfduringload Savoy, Kevin - Williston, ND wrote: Ok so I'm the only one not getting this to work. Maybe I'm doing something wrong. Here is the installation I'm using. Install Fedora Core 4 and do all the updates through yum. Then I install zdlib-devel, openssl-devel, newt-devel, gcc, gcc-c++ and then mysql and perl-DBD-MySQL all using yum install. Am I missing something? Something I'm installing I shouldn't be? After doing the Asterisk-Addons with ./configure, make and then make install as it instructs, the two files below do NOT exist anywhere on my system. Can I compile these manually? If so how? Help? What does ./configure say for MySQL? Should be two lines: checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... yes If it's like the above then the CDR module should compile. -- Joshua Colp 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 ___ --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] HT-496 updates
Each time I tried to update the firmware on two HT-496 boxes I got Timeout error sending .bin from (192.168.1.94), 0 bytes Then, Transmit error while sending to 192.168.1.94. The connection is reset by the remote side. I tried on my LAN, and at last with a crossover cable between my notebook and the HT-496. Any ideas? Carlos Alperin ___ --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] Music On Hold Between Servers
Can someone tell me how Asterisk handles music-on-hold between servers? Documentation for this is non-existent. Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2. 1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both? 2. Is the situation any different if the 'trunk' between pbx1 and pbx2 is SIP or IAX? Thanks, Doug. ___ --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] 1.4.0, IMAP and Dovecot
Tzafrir wrote:
On Thu, Dec 28, 2006 at 12:35:57PM +1300, Ray Jackson wrote:
Dan,
I have IMAP support working now with Courier IMAP. Since Courier
(and
probably Dovecot) do not support a single authuser connection that
may
access any mailbox, you have to omit the 'authuser' and
'authpassword'
settings in voicemail.conf and then add the username/password login
per
extension... e.g.
Are you sure that this is an explicit support in the mail server?
Here's what Mark Crispin (the author of both UW-imapd and c-client)
wrote recently:
| Does UW-IMAP have an admin user? If so, where is it configured?
|
| It's hidden in the release notes file.
|
| Any user who is in a UNIX group called mailadm has administrator
| rights in UW imapd and ipop3d. Administrator rights are the right
to
| log in as any other user.
|
| For c-client based client programs (mailutil, Pine, Alpine, etc.),
the
| /authuser flag is used by the mail administrator. For example the
| mailbox name specifier:
|
| {imap.example.com/authuser=fred/user=joe}INBOX
|
| will open a connection to imap.example.com and log in as user
| joe using user fred's password, and then open joe's INBOX. This
assumes
| that user fred is in group mailadm on the imap.example.com.
Fedora does not have a mailadm group, or at least did not when I
installed
this system, but this was yet another good clue.
So can you do this trick manually? authenticate as one user and read
another user's mailbox?
Here's an example with root and pre-authentication. I figure that some
tricks with pam and such will get you further:
[EMAIL PROTECTED] MAIL=maildir:/home/tzafrir/Maildir /usr/lib/dovecot/imap
* PREAUTH [CAPABILITY IMAP4rev1 SORT THREAD=REFERENCES MULTIAPPEND
* UNSELECT LITERAL+ IDLE CHILDREN LISTEXT LIST-SUBSCRIBED NAMESPACE]
* Logged in as root
1 list *
* LIST (\HasNoChildren) . INBOX
1 OK List completed.
2 select INBOX
* FLAGS (\Answered \Flagged \Deleted \Seen \Draft)
* OK [PERMANENTFLAGS (\Answered \Flagged \Deleted \Seen \Draft \*)]
* Flags permitted.
* 0 EXISTS
* 0 RECENT
* OK [UIDVALIDITY 1161851409] UIDs valid
* OK [UIDNEXT 1] Predicted next UID
2 OK [READ-WRITE] Select completed.
* logout
* BYE Logging out
* OK Logout completed.
I played around with mtest in the c-client package somemore and found
that Dovecot does not permit root logons period. So I added a basic
Unprivilaged account and set the authuser/authpassword in voicemail.conf
to use that account. Now it works, except maybe the expunge bit, which
is likely a config issue.
This is dovecot 0.99.14 on Debian Sarge. Note that I actually don't
use
that imap mailbox normally.
Thanks to everyone for the help. I'm looking forward to some of the
IMAP
enhancements listed in the bugtracker. My mother-in-law has an
extension
on my system, but no mailbox, so I would love to have her extension use
legacy VM and my wife and I get the new IMAP storage.
Dan
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Re: [asterisk-users] MixMonitor and Queues
Recompiled Asterisk after installing sox and it's still not merging the
two streams into a single recorded file. What am I doing wrong?
Jay
Jay Moore wrote:
Ed,
Thanks for the help. One more question, however. Everything is working
fine with the exception of sox joining the calls. I have sox installed
and monitor-join set to yes in both queues.conf and agents.conf
I installed sox after I installed Asterisk. Do I need to recompile
Asterisk for it to work with sox?
This is the last hurdle I need to overcome (I hope) before I can use my
Asterisk box in a live situation. Any help would be much appreciated.
Regards,
Jay
Ed Nuñez wrote:
In queues.conf you must have the following under the queues you want
to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
this will join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls;this is the path where call
will be recorded.
That's all
If you want to change the file name place this in your extensions.conf
on a line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones
for 15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once
the call comes out of queue (no point in recording hold music, unless
I want to hear people mumble about how lousy a company we are for
placing them on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I
ask is that I'd like to put the receiving extension in the name of the
file that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to
record the call when it's pulled out of either queue using
MixMonitor. Bonus points if I can determine the answering extension
before MixMonitor starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
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[asterisk-users] vzaphfc?
Hi list! I'm totally fed up with bristuff (or it's instability with a simple HFC-S card), 2 out of 3 times when people try to call they get the information tone that the number is not connected. I would like to try vzaphfc and I am looking for information on it. From previous posts I found that the only place where the sources seem to be maintained and available is at the debian site which I found here : http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/ But I couldn't find any place where I could download a tarball. Is vzaphfc an inplace replacement of the zaptel of bristuff? Or something separate? 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
Re: [asterisk-users] Checking voicemail from outside
Rob,
Interestingly enough, I'm using that same sample macro, and that line is
indeed in there, yet when I hit *, I hear the tone to leave a message.
Any ideas?
Phil
Phil,
Add this to your extensions (I have mine in a macro)
exten = a,1,VoicemailMain(${ARG1}); If they press *, send to
Voicemail
so it should look like...
exten = s,1,Dial(${ARG2},13,${ARG3})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) exten =
s-BUSY,1,Voicemail(b${ARG1}) exten = a,1,VoicemailMain(${ARG1})
I have a few other things in there as well, but those are the lines that
should do what you want. When you press *, you are prompted for a
password.
Rob
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[asterisk-users] 1.4 - G729 - Have License - No path to translate from Zap to IAX2
Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756] chan_zap.c: Requested indication 3 on channel Zap/1-1 [Dec 28 21:06:02] WARNING[1734] chan_iax2.c: Received mini frame before first full voice frame . . [Dec 28 21:06:02] WARNING[1736] chan_iax2.c: Received mini frame before first full voice frame [Dec 28 21:06:02] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 answered Zap/1-1 [Dec 28 21:06:02] WARNING[1756] channel.c: No path to translate from Zap/1-1(68) to IAX2/VoIPRakyat-2(256) [Dec 28 21:06:02] WARNING[1756] app_dial.c: Had to drop call because I couldn't make Zap/1-1 compatible with IAX2/VoIPRakyat-2 I just upgraded to SVN-branch-1.4-r49020M, but doesn't help. I am using TDM400P with one FXO and one FXS. Initially I just compiled and loaded zaptel and wctdm modules. Then I tried to compile and load ztd-eth, ztd-loc, ztdummy, ztdynamic and zttranscode modules as well just to make sure, but that does not help either. I have no issue at all using any other codecs on IAX. There are some threads on this mailing list for similar issue, but mostly pointed out to G729 license. I have one as below: [Dec 28 21:02:52] VERBOSE[1440] logger.c: == G.729 Host-ID: ... [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Found license 'G729-' providing 1 channels [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Found total of 1 G.729 licenses [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Registered translator 'g729tolin' from format g729 to slin, cost 6 There must be something basic that I missed, maybe the new 1.4 parameters, but I don't know which ones. So please help me out. Thanks a lot in advance. Cheers, Anto ___ --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] [OT] Wifi SIP phones - LinkSys WIP330
Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne . ___ --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] one way rtp stream (Sent alwax to 127.0.0.1)
Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI rtp debug RTP Debugging Enabled -- Executing Dial(SIP/xlite-007918f0, SIP/snom) in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered SIP/xlite-007918f0 -- Attempting native bridge of SIP/xlite-007918f0 and SIP/snom-00797110 Got RTP packet from 192.168.100.70:50002 (type 0, seq 6022, ts 32652224, len 160) Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49874, ts 64, len 160) Got RTP packet from 192.168.100.20:17548 (type 0, seq 6911, ts 1973300, len 160)Sent RTP packet to 192.168.100.70:50002 (type 0, seq 28956, ts 16, len 160) Got RTP packet from 192.168.100.70:50002 (type 0, seq 6023, ts 32652544, len 160) Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49875, ts 384, len 160) *CLI sip show peers Name/username HostDyn Nat ACL Port Status snom/snom 192.168.100.70 D 2051 Unmonitored xlite/xlite192.168.100.20 D 11420Unmonitored 2 sip peers [2 online , 0 offline] ___ --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] Re: [OT] Wifi SIP phones - LinkSys WIP330
I bought a WIP300 to test and it was aweful. It would either not register a keypress or register it twice. It would also freeze up few minutes at a time. It looks like the WIP330 has a new keypad, so maybe that problem is gone. The WIP300 worked with asterisk, but I can not recall the quality at this point. -- -- Steven http://www.glimasoutheast.org Wayne [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne . ___ --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] MixMonitor and Queues
Jay,
I had a similar issue recently... My filename had more than one .
(dot / period)
and the sox version I was using failed to mix files in such conditions...
If that is your case, try:
- Using a filename with no .
- Upgrade sox to the latest version which fixes the funny behaviour
Cheers,
--
Ex Vito
On 12/28/06, Jay Moore [EMAIL PROTECTED] wrote:
Recompiled Asterisk after installing sox and it's still not merging the
two streams into a single recorded file. What am I doing wrong?
Jay
Jay Moore wrote:
Ed,
Thanks for the help. One more question, however. Everything is working
fine with the exception of sox joining the calls. I have sox installed
and monitor-join set to yes in both queues.conf and agents.conf
I installed sox after I installed Asterisk. Do I need to recompile
Asterisk for it to work with sox?
This is the last hurdle I need to overcome (I hope) before I can use my
Asterisk box in a live situation. Any help would be much appreciated.
Regards,
Jay
Ed Nuñez wrote:
In queues.conf you must have the following under the queues you want
to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
this will join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls;this is the path where call
will be recorded.
That's all
If you want to change the file name place this in your extensions.conf
on a line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones
for 15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once
the call comes out of queue (no point in recording hold music, unless
I want to hear people mumble about how lousy a company we are for
placing them on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I
ask is that I'd like to put the receiving extension in the name of the
file that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to
record the call when it's pulled out of either queue using
MixMonitor. Bonus points if I can determine the answering extension
before MixMonitor starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
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RE: [asterisk-users] Re: [OT] Wifi SIP phones - LinkSys WIP330
I recommend the hitachi wifi phones for use with asterisk. Bryan M. Johns Partner Shelton Johns Technology Group Office: (678) 248-2637 X: 1500 Direct: (678) 229-1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Steven [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 12/28/2006 4:30 PM Subject: [asterisk-users] Re: [OT] Wifi SIP phones - LinkSys WIP330 I bought a WIP300 to test and it was aweful. It would either not register a keypress or register it twice. It would also freeze up few minutes at a time. It looks like the WIP330 has a new keypad, so maybe that problem is gone. The WIP300 worked with asterisk, but I can not recall the quality at this point. -- -- Steven http://www.glimasoutheast.org Wayne [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne . ___ --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] ONE WAY VOICE ONLY IN ASTERISK
Try setting in sip.conf: nat=route This tells asterisk to send all responses back to where the inquiry came from rather then from the info contained in the sip packet. Good luck, Mark Coccimiglio IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com Elpidio Ramos wrote: This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux: My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC) I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces. I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop. This is my sip.conf file: [general] context=ramosoft allowguest=no realm=ramosoft.com bindaddr=0.0.0.0 bindport=5060 srvlookup=yes pedantic=yes tos=184 tos=lowdelay maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=ilbc allow=gsm musicclass=default language=es relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 useragent=RamoSoftPBX regcontext=ramosoft localnet=10.10.10.0/255.255.255.0 rtcachefriends=yes [authentication] [311] type=friend regexten=311 username=311 secret=311 callerid=Elpidio Ramos 311 host=dynamic nat=yes canreinvite=no Is there anything I am missing here to get two way voice? Thank you in advance all ___ --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] one way rtp stream (Sent alwax to 127.0.0.1)
Found problem xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't know how to change this at xlite venus*CLI -- SIP read from 192.168.100.20:60726: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK02d4cc64;rport=5060 Contact: sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea To: sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea;tag=7b512144 From: Hans-Juergen Brandsip:[EMAIL PROTECTED];tag=as4530bf3b Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 179 v=0 o=- 5 2 IN IP4 127.0.0.1 s=CounterPath X-Lite 3.0 c=IN IP4 127.0.0.1 t=0 0 m=audio 59050 RTP/AVP 0 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 127.0.0.1:59050 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea set_destination: Parsing sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea for address/port to send to set_destination: set destination to 192.168.100.20, port 60726 Transmitting (no NAT) to 192.168.100.20:60726: ACK sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea SIP/2.0 Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK42575a4c;rport From: Hans-Juergen Brand sip:[EMAIL PROTECTED];tag=as4530bf3b To: sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea;tag=7b512144 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Original-Nachricht Datum: Thu, 28 Dec 2006 22:30:24 +0100 Von: Hans-Jürgen Brand [EMAIL PROTECTED] An: asterisk-users@lists.digium.com Betreff: [asterisk-users] one way rtp stream (Sent alwax to 127.0.0.1) Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI rtp debug RTP Debugging Enabled -- Executing Dial(SIP/xlite-007918f0, SIP/snom) in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered SIP/xlite-007918f0 -- Attempting native bridge of SIP/xlite-007918f0 and SIP/snom-00797110 Got RTP packet from 192.168.100.70:50002 (type 0, seq 6022, ts 32652224, len 160) Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49874, ts 64, len 160) Got RTP packet from 192.168.100.20:17548 (type 0, seq 6911, ts 1973300, len 160)Sent RTP packet to 192.168.100.70:50002 (type 0, seq 28956, ts 16, len 160) Got RTP packet from 192.168.100.70:50002 (type 0, seq 6023, ts 32652544, len 160) Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49875, ts 384, len 160) *CLI sip show peers Name/username HostDyn Nat ACL Port Status snom/snom 192.168.100.70 D 2051 Unmonitored xlite/xlite192.168.100.20 D 11420 Unmonitored 2 sip peers [2 online , 0 offline] ___ --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] vzaphfc?
On 21:04, Thu 28 Dec 06, Remco Barendse wrote: Hi list! I'm totally fed up with bristuff (or it's instability with a simple HFC-S card), 2 out of 3 times when people try to call they get the information tone that the number is not connected. I would like to try vzaphfc and I am looking for information on it. From previous posts I found that the only place where the sources seem to be maintained and available is at the debian site which I found here : http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/ But I couldn't find any place where I could download a tarball. Is vzaphfc an inplace replacement of the zaptel of bristuff? Or something separate? Remco, When you found out stuff, specially how to make stuff with a simple HFC-S card stable please let me know. We are not deploying them cards anymore because we never get it stable. Real simple setups can be done with a FRITZ!PCI card, but I really prefer the quadbri cards for ISDN2 -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Re: [OT] Wifi SIP phones - LinkSys WIP330
On 12/28/06, Bryan M. Johns [EMAIL PROTECTED] wrote: I recommend the hitachi wifi phones for use with asterisk. Bryan M. Johns Partner Shelton Johns Technology Group Office: (678) 248-2637 X: 1500 Direct: (678) 229-1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Steven [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 12/28/2006 4:30 PM Subject: [asterisk-users] Re: [OT] Wifi SIP phones - LinkSys WIP330 I bought a WIP300 to test and it was aweful. It would either not register a keypress or register it twice. It would also freeze up few minutes at a time. It looks like the WIP330 has a new keypad, so maybe that problem is gone. The WIP300 worked with asterisk, but I can not recall the quality at this point. -- -- Steven http://www.glimasoutheast.org Wayne [EMAIL PROTECTED] wrote in message news: [EMAIL PROTECTED] Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne . Funny you should mention this. I just pulled a WIP300 out of a box about 5 minutes ago to test it. First impression: The speaker sucks. All calls sound like there's an ill-tuned radio in the background, with some kind of squealing always present. Also a fair amount of static. The AP is about 5 feet away, so I don't think its the connectivity. I'm not giving up on this phone yet though. Will report back with more if this topic still lives when I'm finished. ___ --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] [OT] Wifi SIP phones - LinkSys WIP330
Hi Wayne, I was a very lucky guy this christmas, and received a D-Link DPH-540. Despite the very first gen feel of the phone, I have been very impressed so far. You are correct in thinking that it can act as an extension external to your network. So long as the place you're in has a decent router, it shouldn't be a problem. I have tested the phone within my local network, as well as on three other wifi networks that my friends gave me the WEP keys for, and I was able to register fine, as well make and receive calls without issue. On one network, I needed to turn the registration refresh down to 90 seconds (down from one hour) to keep the NAT hole open (but I have to do that with my Polycom 501 at the office too). I set the phone to use G729 (to lower bandwidth usage), and I've found the quality to be great. Depending on where I was, there was a slight delay, but that's typical of any IP phone outside the local net if the router is QoSing VoIP or the net connection isn't up to snuff. The only negative things I have to say about the phone are: 1) You can only store 6 network profiles. I can think of 5 off the top of my head that I visit frequently. If the 6th is left unused for open APs, what happens when I find a sixth wifi enabled venue that I visit? Hopefully this is an artificial limit that will be upped with a firmware upgrade. 2) The refresh rate is _terrible_. It's not really an issue since you're generally not looking at the screen except for dialing, but it would be nice to see some type of fluid refresh. 3) Data entry is rough. There are only two input modes: text or numeric. The text mode defaults to uppercase characters, and if you want to enter a lowercase character, you have to cycle through all the uppercase characters on a key before you reach the lowercase ones. For example, a lowercase a takes four taps of the 2 key. WEP keys are case-insensitive, so that doesn't matter, but phone book entries are a nightmare. The only saving grace for this is that you can access the phone via a web interface and edit your phone book from there. I've found that I get a number from someone, type their name quickly in uppercase and then fix it later via PC when I'm connected at home. Cheers, Alex On 12/28/06, Wayne [EMAIL PROTECTED] wrote: Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne . ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --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] The Good, Bad and Scam VoIP Providers
I agree, he sent me one off list, too - making all kinds of allegations of my sexual preferences. I sent him a link to AA, DrPhil, National Institute of Mental Health and suggested he get some help. On 12/28/06, Alex Robar [EMAIL PROTECTED] wrote: Of course everyone is allowed to use VoIP... Asterisk is open! I think Dovid's point was more that this guy's website says he buys and sells precious metals and other random items, his postings on this list indicate that he installs PBXes and resells VoIP services, and then his private e-mails say that he's a PI. The PI thing sounds just like him trying to get those who attacked him to back off. Alex On 12/28/06, Kevin Walsh [EMAIL PROTECTED] wrote: Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/ [EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://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] mIDN question
Hi, I have switched a while back from chan_capi to chan_misdn. When the number is dialed and the phone is then picked up everything works just fine. Some users however FIRST pick up the phone and then start to dial... I did not get this to work with misdn. When two digits have been dialed, asterisk sees the extension as complete and does not wait for further digits. I am using an midsn NT port that feeds into following dialplan context: [intern] exten = _X.,1,Macro(dial) How is this done properly with misdn? Thanks, Arik PS: I am using the following options in misdn.conf (basically they are the defaults): [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=misdn language=de musicclass=default senddtmf=yes far_alerting=no allowed_bearers=all nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=yes need_more_infos=no method=standard dialplan=0 localdialplan=4 cpndialplan=0 early_bconnect=no incoming_early_audio=no nodialtone=no presentation=-1 screen=-1 jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no hold_allowed=yes [intern] ports=1 callgroup=1 pickupgroup=1 context=intern [extern] ports=2 context=extern msns=* echocancel=128 ___ --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] MixMonitor and Queues
Well I'll be. That fixed it nicely. I was adding the .gsm extension
myself not realizing that Asterisk did it as well. Removing my addition
fixed the problem.
Thanks a ton!
Jay
Ex Vitorino wrote:
Jay,
I had a similar issue recently... My filename had more than one .
(dot / period)
and the sox version I was using failed to mix files in such conditions...
If that is your case, try:
- Using a filename with no .
- Upgrade sox to the latest version which fixes the funny behaviour
Cheers,
--
Ex Vito
On 12/28/06, Jay Moore [EMAIL PROTECTED] wrote:
Recompiled Asterisk after installing sox and it's still not merging the
two streams into a single recorded file. What am I doing wrong?
Jay
Jay Moore wrote:
Ed,
Thanks for the help. One more question, however. Everything is
working
fine with the exception of sox joining the calls. I have sox installed
and monitor-join set to yes in both queues.conf and agents.conf
I installed sox after I installed Asterisk. Do I need to recompile
Asterisk for it to work with sox?
This is the last hurdle I need to overcome (I hope) before I can use my
Asterisk box in a live situation. Any help would be much appreciated.
Regards,
Jay
Ed Nuñez wrote:
In queues.conf you must have the following under the queues you want
to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
this will join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls;this is the path where call
will be recorded.
That's all
If you want to change the file name place this in your extensions.conf
on a line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay
Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones
for 15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once
the call comes out of queue (no point in recording hold music, unless
I want to hear people mumble about how lousy a company we are for
placing them on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I
ask is that I'd like to put the receiving extension in the name of the
file that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to
record the call when it's pulled out of either queue using
MixMonitor. Bonus points if I can determine the answering extension
before MixMonitor starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
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RE: [asterisk-users] [OT] Wifi SIP phon es - LinkSys WIP330
-Ursprüngliche Nachricht- Von: Wayne [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 28. Dezember 2006 22:20 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330 Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne We tried the Siemens Gigaset SL75 W-LAN in a customer's asterisk installation. Voice quality is superb, standbytime and range are ok, looks really convincing. Pricing about 169.- € Regards Guido ___ --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] vzaphfc?
On Thu, Dec 28, 2006 at 09:04:40PM +0100, Remco Barendse wrote: Hi list! I'm totally fed up with bristuff (or it's instability with a simple HFC-S card), 2 out of 3 times when people try to call they get the information tone that the number is not connected. I would like to try vzaphfc and I am looking for information on it. From previous posts I found that the only place where the sources seem to be maintained and available is at the debian site which I found here : http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/ But I couldn't find any place where I could download a tarball. Is vzaphfc an inplace replacement of the zaptel of bristuff? Or something separate? vzaphfc is not a complete replacement of bristuff. It replies on most of it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI driver for HFC-s-based PCI cards. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] 1.4 Random disconnects
Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? Thanks, Jason Jason Adams Sumo Systems 4694 Cemetery Road Suite 310 Hilliard, OH 43026 Phone | 614.433.9906 ext: 102 Fax | 614.433.9931 E-mail | [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] ___ --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] [OT] Wifi SIP phones - LinkSys WIP330
On Thu, 28 Dec 2006, Wayne wrote: Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been using an UT Starcom F1000G for a while now, and so-far so good. It has a bit of a toy feel to it - monochrome display, but actually, it seems to do what it says it does on the back of the packet, and it's battery life is amazing! (3 days on standby) There is a higher grade model in a clam-shall design with a colour screen, but as far as I could tell (a friend has one) it has exactly the same functionality as the bar one I have. It does occasionally lose contact with the base station, but it also has a (good!) knack of finding open access points (when I've been quite surprised to hear it's connection beep go off in my pocket, and then had the ability to make calls through it to my office * server!) I'm not sure I'm quite ready to recommend it to my paying customers yet, but thats probably because they are using rubbish WiFi systems. (I'm not a fan of WiFi, but after building a few community broadband systems out of it will tolerate it!) Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? The F1000G will talk to a STUN server to get round NAT, so as long as the router that hot-spot is connected to isn't doing any real firewalling, just NAT, it just works ... I've been able to enter WEP and WPA keys into it through the keypad, without too much difficulty - I guess it would be much easier if you were an SMS junkie though (my mobile phones have always been Nokia communicators with a qerty keyboard for sending messages, so I've never really gotten into using the numbers pad to compose text or search the contacts list!) The one down-side is that if you are connecting to an AP that has a web based front-end to let you enter your usenrame/password or credit card details (eg. BT OpenWallet) then you're stuffed as it doesn't have a web browser. Gordon ___ --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] vzaphfc?
On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote: vzaphfc is not a complete replacement of bristuff. It replies on most of it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI driver for HFC-s-based PCI cards. Further, if you're looking for 'something else' re: cheapo ISDN cards, definately give Asterisk 1.4 and mISDN a look - no BRIStuff, no huge patches, no wacky stuff.. all Asterisk-core support that worked really well in the brief time I tested it. The key difference is rather than generating 8000 interrupts per second, the mISDN kernel driver (which itself can be thought of 'isdn4linux' version 2.0) polls the card, leading to much lower system load, and no 'wanted 8 bytes, read 7!' errors from dmesg. Cheers, Gavin. ___ --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] Compiling Zaptel 1.4.0 on SuSE 10.0
Folks, I have been trying to install Zaptel 1.4.0 on my SuSE 10.0 box with kernel 2.6.13-15.12. I have installed the kernel sources and run make cloneconfig and make prepare. I have run ./configure but make linux26 is failing with the following error: hawk:/tmp/zaptel-1.4.0 # make linux26 make[1]: Entering directory `/tmp/zaptel-1.4.0/menuselect' make[2]: Entering directory `/tmp/zaptel-1.4.0/menuselect' make[2]: `menuselect' is up to date. make[2]: Leaving directory `/tmp/zaptel-1.4.0/menuselect' make[1]: Leaving directory `/tmp/zaptel-1.4.0/menuselect' make -C /usr/src/linux SUBDIRS=/tmp/zaptel-1.4.0 modules make[1]: Entering directory `/usr/src/linux-2.6.13-15' CC [M] /tmp/zaptel-1.4.0/ztdummy.o /tmp/zaptel-1.4.0/ztdummy.c: In function âztdummy_rtc_interruptâ: /tmp/zaptel-1.4.0/ztdummy.c:158: error: implicit declaration of function âtasklet_hi_scheduleâ /tmp/zaptel-1.4.0/ztdummy.c: In function âinit_moduleâ: /tmp/zaptel-1.4.0/ztdummy.c:275: error: implicit declaration of function âtasklet_initâ /tmp/zaptel-1.4.0/ztdummy.c: In function âcleanup_moduleâ: /tmp/zaptel-1.4.0/ztdummy.c:315: error: implicit declaration of function âtasklet_disableâ /tmp/zaptel-1.4.0/ztdummy.c:316: error: implicit declaration of function âtasklet_killâ make[2]: *** [/tmp/zaptel-1.4.0/ztdummy.o] Error 1 make[1]: *** [_module_/tmp/zaptel-1.4.0] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.13-15' make: *** [linux26] Error 2 hawk:/tmp/zaptel-1.4.0 # As far as I can tell, it's trying to use functions from softirq.h? softirq was apparently removed from the kernel in 2.6.8, though I might be barking up the wrong tree here. I can't seem to find anything on google to suggest what the problem may be. I have been looking through installation guides but can't see anything I've missed. Does anyone have any idea what may cause this? Thanks, Mike. ___ --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] Error compiling chan_vpb
hello this is the error chan_vpb.cc: In function \u2018void mkbrd(vpb_model_t, int)\u2019: chan_vpb.cc:1530: aviso: la dereferencia de punteros de tipo castigado romper las reglas de alias estricto chan_vpb.cc: In function \u2018ast_channel* vpb_new(vpb_pvt*, ast_channel_state, char*)\u2019: chan_vpb.cc:2671: aviso: comparacin entre expresiones enteras signed y unsigned g++-c -o chan_vpb.o chan_vpb.cc [LD] chan_vpb.o chan_vpb.oo - chan_vpb.so chan_vpb.oo: In function `a_gain_vector': /root/asterisk/asterisk-1.4.0/channels/chan_vpb.cc:2251: multiple definition of `a_gain_vector' chan_vpb.o:chan_vpb.cc:(.text+0x130): first defined here /usr/bin/ld: Warning: size of symbol `a_gain_vector' changed from 157 in chan_vpb.o to 151 in chan_vpb.oo chan_vpb.oo: In function `usecount': /root/asterisk/asterisk-1.4.0/channels/chan_vpb.cc:3043: multiple definition of `usecount' chan_vpb.o:chan_vpb.cc:(.text+0x1ce): first defined here /usr/bin/ld: Warning: size of symbol `usecount' changed from 10 in chan_vpb.o to 22 in chan_vpb.oo chan_vpb.oo: In function `description': /root/asterisk/asterisk-1.4.0/channels/chan_vpb.cc:3048: multiple definition of `description' chan_vpb.o:chan_vpb.cc:(.text+0x1d8): first defined here /usr/bin/ld: Warning: size of symbol `description' changed from 10 in chan_vpb.o to 22 in chan_vpb.oo chan_vpb.oo: In function `key': /root/asterisk/asterisk-1.4.0/channels/chan_vpb.cc:3053: multiple definition of `key' chan_vpb.o:chan_vpb.cc:(.text+0x1e2): first defined here /usr/bin/ld: Warning: size of symbol `key' changed from 10 in chan_vpb.o to 22 in chan_vpb.oo chan_vpb.oo: In function `unload_module': /root/asterisk/asterisk-1.4.0/channels/chan_vpb.cc:2782: multiple definition of `unload_module' chan_vpb.o:chan_vpb.cc:(.text+0x4b98): first defined here /usr/bin/ld: Warning: size of symbol `unload_module' changed from 525 in chan_vpb.o to 532 in chan_vpb.oo chan_vpb.oo:(.data+0x0): multiple definition of `DialToneMap' chan_vpb.o:(.data+0x0): first defined here chan_vpb.oo: In function `load_module': /root/asterisk/asterisk-1.4.0/channels/chan_vpb.cc:2847: multiple definition of `load_module' chan_vpb.o:chan_vpb.cc:(.text+0x4da6): first defined here /usr/bin/ld: Warning: size of symbol `load_module' changed from 3274 in chan_vpb.o to 3926 in chan_vpb.oo collect2: ld devolvi el estado de salida 1 make[1]: *** [chan_vpb.so] Error 1 rm chan_vpb.o make: *** [channels] Error 2 hello, if somebody knows like solving this error, to him it will be been thankful. ___ --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] The Good, Bad and Scam VoIP Providers
As if we needed more proof that Bochter was a screw-ball... He's now accused me of being the owner of TRXTel. Not that we needed proof he wasn't actually a PI, but in case anyone had any doubts, read the thread. Alex -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Dec 28, 2006 7:41 PM Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: Alex Robar [EMAIL PROTECTED] There are small minded then there is you Bent Fuck you Your spoof email address is blocked Get a life and stop your scams by hiding.. use a real email address... You are a waste of my time GOOD BYE :-P Best regards, Al Bochter Bochter Serviceshttp://www.BochterServices.com/?t=Email Alex Robar wrote: If you actually wanted to give the information to people, you would have just posted it instead of ranting like a lunatic. Your real problem is that you need attention. Stop being a diva and deal with stuff like this on your own. The bottom line is that if you actually had a case, you would have just proceeded with it and dealt with this privately like any normal, decent person would have done. My gut tells me you have jack shit in terms of evidence, and you were just fired as a customer by Brent for pulling shit like this... Something I would certainly agree with him on if that's what he did. I'll bet this never moves forward and we'll never hear anything about any action you've taken. In fact, I'll bet we'll see the inverse - That TRXTel has sued you for libel for attempting to defame them in public. And FYI, I actually did answer your question, you just didn't read my response... Something quite common in your responses, it seem. Alex On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex But if you READ the posts. I replied to all OFF THE LIST So that is YOUR POINT... They posted my replys That were off the list to the list I blocked the other two jackasses on the server to stop there pointless messages. They can't send any messages to any users at any domains on my servers. The same as we are talking OFF THE LIST // The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. What you need to do is check into the PERSON (*Thats one owner*) that is around 28 years I have a list of 32 others that were scammed by bent Ask me for the links on textel no one as asked for the links.. The point is I am not going to waste any more of my time on the ones like you that don't what the information on the truth. *By the way you never answered my question Do you want to be scammed and lose your money???* New question?? What is unlimited use So your replys are pointless Best regards, Al Bochter Bochter Serviceshttp://www.BochterServices.com/?t=Email Alex Robar wrote: The POINT that you keep whining and complaining about so much, is that you're trying to bully and scare people into ceasing their posts that reflect negatively on you. The original points of your post are not what anyone is focusing on anymore - YOU moved the points away from that by insulting people. Everyone else who is off the point is simply responding to you. The issue here is not that anyone LIKES to be scammed... But that you've insulted valuable, respected members of the Asterisk community simply because of a bad experience you had. To post a complaint is one thing, to rip into someone the way you did is quite another. The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. Alex Robar On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex This is off the list. The point is that I don't like scammers. The ones that tried to attacked are some of the scammers that I am dealing with. Do you like to get scammed out of your money? And what is the point of I am a PI or not. Thats not the point of my message or the subject So if you like to get scammed then there is no point to a reply to this message. Only if you want some links to the sites where you will lose your money... ;-) Hope you have great day! Best regards, Al Bochter Bochter Serviceshttp://www.BochterServices.com/?t=Email Alex Robar wrote: Of course everyone is allowed to use VoIP... Asterisk is open! I think Dovid's point was more that this guy's website says he buys and sells precious metals and other random items, his postings on this list indicate that he installs PBXes and resells VoIP services, and then his private e-mails say that he's a PI. The PI thing sounds just like him trying to get those who attacked him to back off. Alex On 12/28/06, Kevin Walsh [EMAIL PROTECTED] wrote: Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
Here's what he sent me after I told him to shut the up. I kind of wonder if he's just trying to generate traffic at certain sites and it's going to generate ad revenue for him in some lame scheme. Oh well: So you are one of the scammers you are dog shit Good bye you are now blocked like Steve is You are a want to be some one like Bent! Are you in bed with him? Most be I guess you two are good butt buddys :-D :-P Get a life asshole and stop trying to become a geek Your site is slow and looks link shit my dog could do better and he can't type. Get it on a real hosting and get it off your cable/DSL Internet connection You don't have the brain power. 1st graders have more than you do. If you don't want the links to scams then you can't handle the truth Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email http://www.bochterservices.com/?t=Email On 12/28/06, Alex Robar [EMAIL PROTECTED] wrote: As if we needed more proof that Bochter was a screw-ball... He's now accused me of being the owner of TRXTel. Not that we needed proof he wasn't actually a PI, but in case anyone had any doubts, read the thread. Alex -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Dec 28, 2006 7:41 PM Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: Alex Robar [EMAIL PROTECTED] There are small minded then there is you Bent Fuck you Your spoof email address is blocked Get a life and stop your scams by hiding.. use a real email address... You are a waste of my time GOOD BYE :-P Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Alex Robar wrote: If you actually wanted to give the information to people, you would have just posted it instead of ranting like a lunatic. Your real problem is that you need attention. Stop being a diva and deal with stuff like this on your own. The bottom line is that if you actually had a case, you would have just proceeded with it and dealt with this privately like any normal, decent person would have done. My gut tells me you have jack shit in terms of evidence, and you were just fired as a customer by Brent for pulling shit like this... Something I would certainly agree with him on if that's what he did. I'll bet this never moves forward and we'll never hear anything about any action you've taken. In fact, I'll bet we'll see the inverse - That TRXTel has sued you for libel for attempting to defame them in public. And FYI, I actually did answer your question, you just didn't read my response... Something quite common in your responses, it seem. Alex On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex But if you READ the posts. I replied to all OFF THE LIST So that is YOUR POINT... They posted my replys That were off the list to the list I blocked the other two jackasses on the server to stop there pointless messages. They can't send any messages to any users at any domains on my servers. The same as we are talking OFF THE LIST // The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. What you need to do is check into the PERSON (*Thats one owner*) that is around 28 years I have a list of 32 others that were scammed by bent Ask me for the links on textel no one as asked for the links.. The point is I am not going to waste any more of my time on the ones like you that don't what the information on the truth. *By the way you never answered my question Do you want to be scammed and lose your money???* New question?? What is unlimited use So your replys are pointless Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Alex Robar wrote: The POINT that you keep whining and complaining about so much, is that you're trying to bully and scare people into ceasing their posts that reflect negatively on you. The original points of your post are not what anyone is focusing on anymore - YOU moved the points away from that by insulting people. Everyone else who is off the point is simply responding to you. The issue here is not that anyone LIKES to be scammed... But that you've insulted valuable, respected members of the Asterisk community simply because of a bad experience you had. To post a complaint is one thing, to rip into someone the way you did is quite another. The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. Alex Robar On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex This is off the list. The point is that I don't like scammers. The ones that tried to attacked are some of the scammers that I am dealing with. Do you like to get scammed out of your money? And what is the
Re: [asterisk-users] Checking voicemail from outside
You could be using an older version of Asterisk that doesn't support it?
On 12/28/06, Phil Finkler [EMAIL PROTECTED] wrote:
Rob,
Interestingly enough, I'm using that same sample macro, and that line is
indeed in there, yet when I hit *, I hear the tone to leave a message. Any
ideas?
Phil
Phil,
Add this to your extensions (I have mine in a macro)
exten = a,1,VoicemailMain(${ARG1}); If they press
*, send to
Voicemail
so it should look like...
exten = s,1,Dial(${ARG2},13,${ARG3})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) exten =
s-BUSY,1,Voicemail(b${ARG1}) exten = a,1,VoicemailMain(${ARG1})
I have a few other things in there as well, but those are the lines that
should do what you want. When you press *, you are prompted for a password.
Rob
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Snap - A desktop user interface for Asterisk
www.snapanumber.com
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[asterisk-users] TE110P with Qsig
Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, 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. = Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 Dec 28 21:31:57 WARNING[5484]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 1: Red Alarm Dec 28 21:31:57 WARNING[5485]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 1 Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 2: Red Alarm Dec 28 21:31:57 WARNING[5485]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 2 Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 3: Red Alarm Dec 28 21:31:57 WARNING[5485]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 3 Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 4: Red Alarm Dec 28 21:31:57 WARNING[5485]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 4 Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 5: Red Alarm Dec 28 21:31:57 WARNING[5485]: chan_zap.c:1435
RE: [asterisk-users] 1.4 and unicall
I don't think if somebody making upgrades for the unicall in accordance to the latest version of Asterisk. The latest patches of unicall and MFCR2 that I saw is still for Asterisk ver. 1.2.0. Haven't see any patches for latest version yet. This what making me afraid of going to upgrade our Asterisk, I am using MFCR2 as well with Asterisk 1.2.12 without any problem. I hope there will be version of upgrades that it won't delete unicall libraries and its dependencies. Rgds. Angel Anton Krall [EMAIL PROTECTED] wrote: I hope so, he is the only guy working on mfcr2 right now. I have unicall working on 1.2 perfectly but if there will be no unicall support for 1.4, that would be a show stopper unless we use a mfcr2 converter... anybody knows any? Something that can convert mfcr2 to pri? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Barzilai Spinak |Sent: Thursday, December 28, 2006 8:26 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] 1.4 and unicall | |I asked the same a while ago, without any kind of conclusive answer. |But you have to consider that these are special dates |I just spent all night studying/modifying mfcr2.c to my needs but |I've never looked at the unicall code or the asterisk channel API. |With respect to MFC/R2, and according to what it saw, it seems fairly |complete on the incoming part of the protocol, but the outgoing logic is |kind of crude. |I wonder if Steve Underwood is still actively working on it. | |BarZ | |Anton Krall wrote: | No update on unicall and 1.4? | | |-Original Message- | |From: [EMAIL PROTECTED] [mailto:asterisk-users- | |[EMAIL PROTECTED] On Behalf Of Anton Krall | |Sent: Tuesday, December 26, 2006 6:15 AM | |To: asterisk-users@lists.digium.com | |Subject: [asterisk-users] 1.4 and unicall | | | |Guys, anybody knows if 1.4 has support for unicall or if/which version of | |unicall will compile on 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 | | | | ___ | --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 BEGIN:VCARD VERSION:2.1 X-MS-SIGNATURE:YES N;LANGUAGE=en-us:Krall;Anton FN:Anton Krall ORG:Intruder Consulting TITLE:A Division of IntruderEnterprises S.A. de C.V. TEL;WORK;VOICE:+52 (55) 5781-5112 x 201 TEL;WORK;VOICE:+52 (55) 5985-2430 x 201 X-MS-OL-DEFAULT-POSTAL-ADDRESS:0 URL;WORK:http://www.intruder.com.mx EMAIL;PREF;INTERNET:[EMAIL PROTECTED] PHOTO;TYPE=JPEG;ENCODING=BASE64: /9j/4AAQSkZJRgABAQEAYABgAAD/2wBDAAYEBQYFBAYGBQYHBwYIChAKCgkJChQODwwQFxQY GBcUFhYaHSUfGhsjHBYWICwgIyYnKSopGR8tMC0oMCUoKSj/2wBDAQcHBwoIChMKChMoGhYa KCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCgoKCj/wAAR CAAXAEgDASIAAhEBAxEB/8QAHwAAAQUBAQEBAQEAAAECAwQFBgcICQoL/8QAtRAA AgEDAwIEAwUFBAQAAAF9AQIDAAQRBRIhMUEGE1FhByJxFDKBkaEII0KxwRVS0fAkM2JyggkK FhcYGRolJicoKSo0NTY3ODk6Q0RFRkdISUpTVFVWV1hZWmNkZWZnaGlqc3R1dnd4eXqDhIWG h4iJipKTlJWWl5iZmqKjpKWmp6ipqrKztLW2t7i5usLDxMXGx8jJytLT1NXW19jZ2uHi4+Tl 5ufo6erx8vP09fb3+Pn6/8QAHwEAAwEBAQEBAQEBAQECAwQFBgcICQoL/8QAtREA AgECBAQDBAcFBAQAAQJ3AAECAxEEBSExBhJBUQdhcRMiMoEIFEKRobHBCSMzUvAVYnLRChYk NOEl8RcYGRomJygpKjU2Nzg5OkNERUZHSElKU1RVVldYWVpjZGVmZ2hpanN0dXZ3eHl6goOE hYaHiImKkpOUlZaXmJmaoqOkpaanqKmqsrO0tba3uLm6wsPExcbHyMnK0tPU1dbX2Nna4uPk 5ebn6Onq8vP09fb3+Pn6/9oADAMBAAIRAxEAPwDI8HeCb3xFHJclhb2MY+adumfQDua62b4Q iBGluNWEcOPlZreQZJ/4DXS/DAXn/CBxS6fArXscc5gbPOflGfrgmun8NNfr4dt5vESs2oG2 lRNxy7ZZcYHryfwzXo4nGVouTi0knY+ew+CoyhHmV21e58/XvhaWz1XWLGZpC+n27TkxoGBw QOckYHPXn6VFdeDdftbV7i4050iRFkbMiblRsYYrnIXkc4xXWeJNRFp408Xosc9217YyW6mF d20llO5vQDFUdQ8TxT6pr2pi0vBaalpqafCzLwHEcanJzjGUbpXoxqVGk/66HnzpUU2n3f6/ 8D7zO1D4f67bao9lbQJdukEdw7xyKFRXAPJJ4wTjJ9M9Kxl0DVHu7a1Wzcz3MJuIUBHzxgMS w56YVj+FdnrevxXMesW8FjqaX2oaba27RPDjyzDsBPXJUhOuO9LpPijRoDo+p3UOp/bNP017 ARxxqYmyroH3k5/j6Y696FUqct2v6t/mEqNFysnZf8H07amJ4Z8CarrU1o0sRtbK4jeVZmZS 2xVJ3CMsGKkjGcY5rNs/CutXmmrfW1i727Kzod6hnVfvMqE7mA55ANdvYa7psOraXq9/b6ol /YW/9lNBEiGFpVRkBD7hjg5K4696bpnjOC30XSrqSC9huNNtjZqYrOFkkYbtpEzAsn3uQAf1 pOpVvov61/4BSo0LJN/1p/wdNzg9Q0DU9P0221C9tWhtLkBoXZl+cEZBAznGO+KKveM9Qa/b Rw1vcQfZtOhtsTLt3Fc5ZfbNFdEG2rs46qjGVo7Gh4Z8e3uhaetpHbW9wiZ8syru2Z64BrRn +KmqPaslva2ttcFCnnxIFYA9en86KKyeGpSlzOKuaRxdaK5VLQ4qHVLiOaeRisrTDD+YM8g5 B+oIpG1KdtPSzITy0xhsfMQCzAfTLMfxoorblRjzPuWItevY74XeY2l2svK8EM5c5APPzMeO
[asterisk-users] Re: asterisk-users Digest, Vol 29, Issue 114
Can someone tell me how Asterisk handles music-on-hold between servers? Documentation for this is non-existent. Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2. 1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both? 2. Is the situation any different if the 'trunk' between pbx1 and pbx2 is SIP or IAX? I'm using IAX trunk between the servers. It tested fine. After switching the [default] mode=files instead of mode=quietmp3 (restart the pbx, don't just reload res_musiconhold.so), I could call from pbx1 to pbx2, from pbx2 put the call on hold, pbx2 would play the MOH correctly. Call pbx2 to pbx1, put the call on hold from pbx1 and pbx1 played the MOH correctly. Didn't see any issues past mode=files. -- JR Richardson Engineering for the Masses ___ --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] Checking voicemail from outside
Or more likely the tone may not be getting to asterisk. What FXO are you using? External FXO's like the SPA3000 often need to be set to 'inband' DTMF - both in sip.config and in the device's config and be sure to restart Asterisk after doing this.. Easiest way to test this is to call yourself from your cell and see if you can hear the DTMF tones on the Asterisk side as you enter them on your cell. If you can not hear them then Asterisk won't decode them! This is also necesssary for outgoing FXO calls to enable use of external IVR's like banking and business voice menus. There is much about this on this list in the past and in Asterisk bug reports. It is not exactly clear where the problem lies but it appears to be a combination of Asterisk and the SPA3000. This might be fixed in version 1.4 but I have not heard any reports as yet. Doug On Thu, 28 Dec 2006, mitcheloc wrote: You could be using an older version of Asterisk that doesn't support it? On 12/28/06, Phil Finkler [EMAIL PROTECTED] wrote: Rob, Interestingly enough, I'm using that same sample macro, and that line is indeed in there, yet when I hit *, I hear the tone to leave a message. Any ideas? Phil ___ --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] Boot load wcfxo does not configure self underUbuntu 6
On Fri, Dec 15, 2006 at 06:32:19PM -0800, Yuan LIU wrote: When booting Ubuntu 6.06.1 (Linux 2.6.15-27-386), wcfxo would load but not configure. I have three ways to manually force wcfxo to configure: 1) ztcfg, 2) modprobe -f wcfxo, or of course 3) unload and reload wcfxo. Each works equally well. The usual confusion about init scripts. Debian's init scripts automatically load the module for this card using coldplug (a run of hotplug when the system starts). However the modprobe of the module fails due to the silly automatic run of ztcfg at module load tme with very stupid modprobe settings. So as a workaround you unload and reload the module. Not smart, and has a potential for races. What is the output of: grep wcfxo /etc/modprobe.d/* zaptel:install wcfxo /sbin/modprobe -s --ignore-install wcfxo $CMDLINE_OPTS /sbin/ztcfg Looks like a correct syntax. Yuan Liu -- Tzafrir Cohen ___ --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] The Good, Bad and Scam VoIP Providers
I am after a long day of work, it felt realy good to laugh a bit. On 12/28/06, Tom Lynn [EMAIL PROTECTED] wrote: Here's what he sent me after I told him to shut the up. I kind of wonder if he's just trying to generate traffic at certain sites and it's going to generate ad revenue for him in some lame scheme. Oh well: So you are one of the scammers you are dog shit Good bye you are now blocked like Steve is You are a want to be some one like Bent! Are you in bed with him? Most be I guess you two are good butt buddys :-D :-P Get a life asshole and stop trying to become a geek Your site is slow and looks link shit my dog could do better and he can't type. Get it on a real hosting and get it off your cable/DSL Internet connection You don't have the brain power. 1st graders have more than you do. If you don't want the links to scams then you can't handle the truth Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email On 12/28/06, Alex Robar [EMAIL PROTECTED] wrote: As if we needed more proof that Bochter was a screw-ball... He's now accused me of being the owner of TRXTel. Not that we needed proof he wasn't actually a PI, but in case anyone had any doubts, read the thread. Alex -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Dec 28, 2006 7:41 PM Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: Alex Robar [EMAIL PROTECTED] There are small minded then there is you Bent Fuck you Your spoof email address is blocked Get a life and stop your scams by hiding.. use a real email address... You are a waste of my time GOOD BYE :-P Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Alex Robar wrote: If you actually wanted to give the information to people, you would have just posted it instead of ranting like a lunatic. Your real problem is that you need attention. Stop being a diva and deal with stuff like this on your own. The bottom line is that if you actually had a case, you would have just proceeded with it and dealt with this privately like any normal, decent person would have done. My gut tells me you have jack shit in terms of evidence, and you were just fired as a customer by Brent for pulling shit like this... Something I would certainly agree with him on if that's what he did. I'll bet this never moves forward and we'll never hear anything about any action you've taken. In fact, I'll bet we'll see the inverse - That TRXTel has sued you for libel for attempting to defame them in public. And FYI, I actually did answer your question, you just didn't read my response... Something quite common in your responses, it seem. Alex On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex But if you READ the posts. I replied to all OFF THE LIST So that is YOUR POINT... They posted my replys That were off the list to the list I blocked the other two jackasses on the server to stop there pointless messages. They can't send any messages to any users at any domains on my servers. The same as we are talking OFF THE LIST // The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. What you need to do is check into the PERSON (Thats one owner) that is around 28 years I have a list of 32 others that were scammed by bent Ask me for the links on textel no one as asked for the links.. The point is I am not going to waste any more of my time on the ones like you that don't what the information on the truth. By the way you never answered my question Do you want to be scammed and lose your money??? New question?? What is unlimited use So your replys are pointless Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Alex Robar wrote: The POINT that you keep whining and complaining about so much, is that you're trying to bully and scare people into ceasing their posts that reflect negatively on you. The original points of your post are not what anyone is focusing on anymore - YOU moved the points away from that by insulting people. Everyone else who is off the point is simply responding to you. The issue here is not that anyone LIKES to be scammed... But that you've insulted valuable, respected members of the Asterisk community simply because of a bad experience you had. To post a complaint is one thing, to rip into someone the way you did is quite another. The way you insulted the owners of TRXTel, not to mention the half a dozen other list members who defended them, was very childish. Alex Robar On 12/28/06, Al Bochter [EMAIL PROTECTED] wrote: Alex This is off the list. The point is that I don't like scammers. The ones that tried to
Re: [asterisk-users] vzaphfc?
On Thu, 28 Dec 2006, Gavin Hamill wrote: On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote: vzaphfc is not a complete replacement of bristuff. It replies on most of it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI driver for HFC-s-based PCI cards. Further, if you're looking for 'something else' re: cheapo ISDN cards, definately give Asterisk 1.4 and mISDN a look - no BRIStuff, no huge patches, no wacky stuff.. all Asterisk-core support that worked really well in the brief time I tested it. The key difference is rather than generating 8000 interrupts per second, the mISDN kernel driver (which itself can be thought of 'isdn4linux' version 2.0) polls the card, leading to much lower system load, and no 'wanted 8 bytes, read 7!' errors from dmesg. Thanks for the tip, I'll have a look at it. The main reason for me to use bristuff is that i don't want to mess mess around downloading and compiling my own kernels. I am just running CentOS 4 boxes with stock CentOS 4 kernels. Everytime I was screwing around with making my own kernels sooner or later I got bitten by screwing up the installation of the kernel and the box wouldn't boot anymore. :) On the wiki I found the manual from BeroNet which looks pretty straightforward but is for Asterisk 1.2 Any differences for Asterisk 1.4? 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
Re: [asterisk-users] vzaphfc?
On Thu, 28 Dec 2006, Michiel van Baak wrote: When you found out stuff, specially how to make stuff with a simple HFC-S card stable please let me know. We are not deploying them cards anymore because we never get it stable. Real simple setups can be done with a FRITZ!PCI card, but I really prefer the quadbri cards for ISDN2 I think I'll try misdn or vzaphfc, if it is too complicated or i'm not satisfied with the results I will simply hook up a good old A/B adapter, convert the ISDN to analog lines and throw in a Digium TDM card. I've pretty much had it with ISDN2 ___ --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] voicemail and ip phones
Hi, In my ip phone is voicemail indicator, and also is a voicemail button (to access to voicemail server and ant to listen voicemail). My question is how to configure this button. In configuration I need to enter URL. What is the syntax of this URL, that IP Phone could fetch this voicemail from asterisk. ___ --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] one way rtp stream (Sent alwax to 127.0.0.1)
On 12/29/06 06:04 Hans-Jürgen Brand said the following: Found problem xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't know how to change this at xlite have you tried nat=yes in sip.conf for the peer ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] [OT] Wifi SIP phones - LinkSys WIP330
Trouble is this (promising) phone is not distributed everywhere, at least, not here in France, yet. I couldn't get any reason from Siemens France. ___ --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: FW: [asterisk-users] cdr_addon_mysql.so did not register itselfduringload
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Savoy, Kevin - Williston, ND wrote: Ok so something is missing. I get the below for those two lines. checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... no I even installed the mysql-devel as Bradley Watkins suggested and still it says no. What do I need to make that say yes? Try a make distclean in the addons directory before doing a ./configure. mysql-devel is definitely what you need. Run ldconfig maybe? You shouldn't need to though. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFlLv9S6d5vy0jeVcRAnFwAJ0YER87bSGBWIpQVWt8zRJQnHhq0wCfV1Z8 Vu+U0ejq3sHAfAlDcBilXUM= =0qw+ -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
