similar to: Queue Warning messages

When I use an eagi script when play a message appear a lot of warning messages, but it play very well I?m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300

Dear all i'm creating an outgoing call to number xxx with this command: http://host:port/mxml?action=Originate&Channel=Local/xxx at to-external &Exten=testDTMF&Context=cRETEUNICA&Priority=1 wich points correctly to this portion of dialplan: [cRETEUNICA] exten => testDTMF,1,Answer exten => testDTMF,n,Read(digito,,1) exten => testDTMF,n,SayDigits(${digito}) The

Hi, I've just built my first asterisk server. Current information: OS Version: Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10 06:50:22 EDT 2007 i686 i686 i386 GNU/Linux Asterisk Build: Asterisk 1.4.11 Asterisk GUI-version Revision: 1479 $ Server Date & TimeZone: Thu Sep 6 02:37:11 EST 2007 I've used the Asterisk GUI for setup with two IP

Guys, when registering I am getting this error message, my question is that if this could be the reason whay I am able to make calls but not to recieve call ? [Apr 3 11:24:31] WARNING[19578]: chan_sip.c:15104 handle_response_register: Got 423 Interval too brief for service +506phonenumber at domain.co.cr@host.ip.addr, minimum is 3600 seconds Thanks -- http://celord.blogspot.com/

I have been getting the following message every time I make a call for the past few months: [Nov 2 13:08:18] WARNING[9859]: file.c:1273 waitstream_core: Unexpected control subclass '-1' Everything seems to be working so I do not know if this is important. I am using Asterisk 1.4.26.1 (upgrading today to .2) with Asterisk Addons 1.4.9, Zaptel 1.4.12.1, Libpri 1.4.10 (upgrading to

Hello guys, anybody here that can help me checking out this xml file, cause I am traying to configure some cisco 7911G phones to asterisk and I can't get it done.... thanks!!!! a paste of the file is here: http://pastebin.ca/1239083 -- http://celord.blogspot.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

This is the core trace (gdb) bt #0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6 #1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828 "default", interpclass=0x0) at res_musiconhold.c:646 #2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 <Address 0x64 out of bounds>, interpclass=0x88 <Address 0x88 out of bounds>) at channel.c:4614 #3

Hello I have the following problem using chan_oh323 I have DialGate 2160 for SystemBas (www.sysbas.com) connected to PSTN (H.323 to FXO/FXS gateway) when i try to make call form one pstn phone to other trough asterisk or when i make call from software h.323 client trough asterisk and this gateway to pstn i have the problem with voice quality. The side that initiated call can be heared clearly,

Hi, I posted this a week ago on r-help but did not get an answer. So I hope that someone here can help me: I want to define some options for my package the user may change. It would be convenient if the changes could be saved when terminating an R session and recovered automatically on the next package load. Is that possible and if yes, is the standard way to implement this? Thanks, Mark

Dear list, I have just started to use roxygen and cannot find out how to specify NOT RUN examples. I know that to include R code examples works via the @examples tag followed by the code. Without roxygen, if I want R code not to be run when processing the .Rd files I would include the \dontrun tag \examples{\dontrun{ ... some example R code} } How can I achieve this using roxygen? Thanks Mark

Hello guys, I am having some problems with calls comming from the PSTN lines, when somebody calls people can't hear me, but I can hear them, every day I have to do a /etc/init.d/asterisk stop && /etc/init.d/dahdi restart to have calls with sound again, wich cli dubug commands can I use to see what is going on, here I have my chan_dahdi.conf and sip.conf, I am using 1.6 Thanks a lot!

Hello, The following patch avoids a warnign message when using hostbased authentication as root and protocol v1 support has been disabled. The case for non-root users has already been addressed, but root follows a different code path. -- Iain Morgan diff --git a/ssh.c b/ssh.c index cceb36e..e32aa0a 100644 --- a/ssh.c +++ b/ssh.c @@ -1242,8 +1242,10 @@ main(int ac, char **av)

Hi, I'm using dovecot 2.1.7 on debian Squeeze with ldap integration. Almost everything is working fine and I would link to thank all developers for this amazing piece of software. But after changing my qmail-ldap to do LDA integration get all the benefits from LDA and sieve, some emails return with failure like this: <sar at mondialline.com.br>: lda(sar): Fatal: Unknown argument: sar

The Asterisk Development Team would like to announce the release of Asterisk 14.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hi! I use dovecot 2.1.7 on Ubuntu 12.10 with fts_solr und decode2text.sh for indexing attachments. This works great in general. Just for one user there is a problem with an unknown bad attachment. I run "doveadm index -A '*'". After a while I receive: doveadm(xyz): Error: fts_solr: Invalid XML input at line 1: mismatched tag doveadm(xyz): Panic: file solr-connection.c: line

Hi, I recently changed my version of asterisk to 11.XX, and I see a waning with h323, with version 1.8 did not have these warning I have connected one avaya ip office 500 h323 with asterisk and the 1.8 version did not have these messages Oct 23 17:20:35] WARNING[7593][C-000000aa]: chan_ooh323.c:1413 ooh323_indicate: Don't know how to indicate condition 33 on ooh323c_60 [Oct 23 17:20:35]

I am trying to register windows messanger with asterisk and it fails. If I try X-lite on the same PC it registers without any problems. MY setup looks like this ---------ROUTER-------- | | | | client1 asterisk Client1 never can connect to asterisk using microsoft messanger. Any ideas. Bart,

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Hello, I have a data set like one below. First, I'd like to replace the empty cells with NA, and then the one immediately above. I could replace NAs with the immediate one, but don't know for the empty cells. index <- which(is.na(data1$year)) while (any(index)) { dummy$data1[index] <- dummy$data1[index - 1] index <- which(is.na(data1$year)) } Second, The id