similar to: SIP Inuse Count Wrong

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. I am getting: -- Executing

I am new to asterisk and have been playing with it for just a few days. The GrandStream BudgeTone-102 equipment I have allows me to do a blind transfer, but I am interested to know how to do something which isn't blind. I have managed to enable the call parking which allows me to send a call to extension 700 for somebody to collect, but it still seems rather awkward. I also have

When you look at 'show translation' you see: Translation times between formats (in milliseconds) but is this the number of milliseconds required to convert 1 packet of data, or the amount of time required to translate 1 second of data? I am assuming it is the time for 1 second. Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq

I had a working configuration whereby an incoming call on an ISDN line would be sent out on the second ISDN line, but since I updated to the latest version of Asterisk I get this error message: WARNING[311315]: File res_parking.c, Line 226 (ast_bridge_call): Bridge failed on channels CAPI[contr1/s]/0 and CAPI[contr1/01624619052]/1 The message comes up as soon as the outgoing call is answered

If you're using perl on RedHat 9 make sure you put this command somewhere in your boot scheme: export LANG=C or at least execute it before running perl scripts. Redhat has EVERYTHING set to LANG=UTF-8 and it screws up all sorts of perl stuff, and several other pre-written programs in other languages too MATT--- -----Original Message----- From: Ray Burkholder

Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been

Hello all, I would like to read the specific line number row of a direct access file (which is stored as a n_row*n_col matrix of elements kind=p) without reading all the preceding lines (i.e 1,2,..,row-1). Is there a function in R that can perform this task? To solve my issue, I tried without to call Fortran from R by doing the following steps: I) I wrote a subroutine in fortran called

Hello All, Thanks to those that responded to my problem of compiling on SUSE 8.2. I was not able to get the compile done so decided to put RedHat 9 on this system. After getting a RedHat supported NIC and RedHat installed, Asterisk compiled cleanly, one SIP phone is connected and voice mail works. No other tests have been run yet. A couple of days ago, Michael Farnworth asked about the smallest

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

Hello, On (03/30/16 16:12), Minchan Kim wrote: [..] > +static int get_zspage_inuse(struct page *first_page) > +{ > + struct zs_meta *m; > + > + VM_BUG_ON_PAGE(!is_first_page(first_page), first_page); > + > + m = (struct zs_meta *)&first_page->mapping; .. > +static void set_zspage_inuse(struct page *first_page, int val) > +{ > + struct zs_meta *m; > + > +

Hello, On (03/30/16 16:12), Minchan Kim wrote: [..] > +static int get_zspage_inuse(struct page *first_page) > +{ > + struct zs_meta *m; > + > + VM_BUG_ON_PAGE(!is_first_page(first_page), first_page); > + > + m = (struct zs_meta *)&first_page->mapping; .. > +static void set_zspage_inuse(struct page *first_page, int val) > +{ > + struct zs_meta *m; > + > +

Hi all, we do have a callcenter system running with 1.4.21.1 - the agents are connected used sip phones. SIP accounts are configured using realtime (sip buddies) - and are configured with call-limit=1. It is operating just fine - but from time to time it does happen that an agent with an active call (inbound or outbound) does start to get a second call offered. I have taken a look at the

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Currently, we store class:fullness into page->mapping. The number of class we can support is 255 and fullness is 4 so (8 + 2 = 10bit) is enough to represent them. Meanwhile, the bits we need to store in-use objects in zspage is that 11bit is enough. For example, If we assume that 64K PAGE_SIZE, class_size 32 which is worst case, class->pages_per_zspage become 1 so the number of objects in

I have recently upgraded to 1.2.7.1 from 1.2.4. I can no longer use "sip show inuse". Below is the output... I know there are current calls: redhat*CLI> sip show inuse * User name In use Limit * Peer name In use Limit Does anyone have an idea why this isn't working? Thanks, bp

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

I have patched astcc.agi with the HUP patch, but it still hangs from time to time. Asterisk SVN-branch-1.2-r25165M built by root @ vpbx on a x86_64 running Linux on 2006-05-07 00:31:09 UTC bye Ronald

Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things