similar to: Problem with call-limit

When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled. (i thought, asterisk itself handles the queues ? ) Here the log: 2007-07-09T15:04:14 YOM04 0 -

http://dovecot.procontrol.fi/test/ - Hopefully OpenSSL is fixed :) - We sometimes produced invalid ENVELOPE with 8bit headers. That could really have broken things (broke it in test12). So, what's left is to make PAM work well..

Hello list, getting warning : *syntax error, unexpected '<token>'* dialplan : exten => pbx,n,Macro(CheckNetworkProblems,${custID}) exten => pbx,n,NoOp(status = ${STATUS}) exten => pbx,n,GoToIf($["${STATUS}"="congestion"]?backup:nocongestion) CLI : [Sep 9 12:27:07] -- Executing [pbx at cust:15] NoOp("SIP/test13-0000002a",

Hi, the function makeARIMA(), designed to construct some state space representation of an ARIMA model, uses a C function called getQ0, which can be found at the end of arima.c in R source files (library stats). getQ0 takes two arguments, phi and theta, and returns the covariance matrix of the state prediction error at time zero. The reference for getQ0 (cited by help(arima)) is:

Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11

Problems with acl in dovecot-1.2.15 was not solved! See my configs below. And I see the new bug - still something wrong with configs: cat dovecot-acl: user=t1 lrwsti #user=oper-olegs lrwsti #user=oper-antona lrwsti user=operdss lr #user=operdss lrwsti User t1 can't create subfolers in inbox. He can only create subfolders in the first levels: Inbox test3 test4 Outbox Sent Trash Test1 - test

Hi, I'm a little surprised, up until 1.4.22 my agents where using an IAX channel to ZoIPer Softphone, however since after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing

Hello! im testing lazy_expunge_only_last_instance here and it seems it works wrong with IMAP MOVE IN: 14 uid move 13 "INBOX" OUT: * OK [COPYUID 1188569061 13 34] Moved UIDs. * 5 EXPUNGE 14 OK Move completed. dovecot.log: 2013-06-14 10:56:06 imap(test13 at mtx.ru): Info: copy from Test: box=INBOX, uid=34, msgid=<1294858169.32435.3.camel at int.office.matrix>, size=996,

https://bugs.freedesktop.org/show_bug.cgi?id=107963 Bug ID: 107963 Summary: kernel rejected pushbuf: Invalid argument Product: xorg Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau Assignee: nouveau at

In doing my testing I''m wondering if maintaining backwards compatability for existing applications is important. The question boils down to this: Are there sufficient applications that are using wxRuby (pre swig) that we should expect to have to support all/most without changes or should we expect that most applications will need to learn the ''new'' ways things

Home again. Fixed SSL checking for Redhat 9, I hope. Could someone test if it actually works now? Also fixed a few compiler warnings. If this thing works, I'll just update the NEWS file and call it 0.99.10. http://dovecot.procontrol.fi/rc/ Near future plans include rewriting parts of index handling. At least .tree file will go, I've a _much_ better idea how to replace it. .data file is

Hello list, I'm using asterisk 1.6.2.11 with realtime SIP (mysql DB). I notice that when the SIP peer registers, the fields 'fullcontact', 'ipaddr', 'port', 'regserver', 'regseconds', 'lastms' are filled with values. But after a while, these fields become empty. Asterisk CLI shows : asterisk*CLI> sip show peers Name/username

hello all, I have asterisk 1.4.26 working on CentOS 5.3 and I have the following problem: - when I restart asterisk all the members of the queue are Invalid. - when I make a call to one of the members, of the queue, and then check the state, it turns to "Not in use" for the called phone, and the queue works fine for that member after. - after doing a module reload of the

Helo List, Sorry I missed the rest of my email in my previous post. Please see below. I'm having an issue using the AND (&) operator evaluation in the code of my dialplan. The dial plan is coded to detect inbound DTMF digits from callers. key "1" is equivalent to "yes" and key "2" is equivalent to "no" in two diferent contexts in the dial plan.

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hi, I'm struggling with a feature in my home phone setup. I have several phones using both SIP and SCCP. What I try to do is to create a dynamic feature that works similar to the blindxfer feature built into Asterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the

Hello all llvm-link merges together the metadata from the IR files being linked together. This means that when linking different libraries together (i.e. multiple source files that have been compiled into a single LLVM IR file) it can be hard or impossible to identify the library boundaries. We're using LLVM to do static analysis of applications (together with their dependent libraries) and

[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members

Hi to all, I need some help, I have an Asterisk Server in a small call center, for inbound calls I setup a Queue in queues.conf and their respective Agents in agents.conf, but when an Agent is calling out and a call is coming from PSTN the call is send to that agents which have a call in progress. How I can fix this in order to have only one call at a time. I think in limitonpeer and call-limit

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a