similar to: Asterisk 1.2.12.1 and snom 360 6.2.3 no audio

Hi, I have * running on a SuSE Email Server II, which uses an older libc-version. Calls to res_ninit et al. from enum.c prevent asterisk from building on that system. Could that possibly be changed to use res_init and res_search in order to provide backward compatibility with these systems ? Thanks, Holger von Ameln

Hi! Since we upgraded to 1.6.1.11, asterisk is only outputing monitored files -in and -out. It is not mixing them in the end. queues.conf has monitor-type=MixMonitor... Would somebody help me debug why it doesn't mix the sounds?? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have scoured the samba archives, and have found a few suggestions to this problem, like setting registry setting on the Windows machine to allow clear text passwords, and making sure permissions are setup correctly in the smb.conf and with the directory to share... but to no avail... Can someone shed some further light on this topic. None of the prior posting's actually had a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from

i'm reading through the samba howto. section 6.2.3 (page 105) shows the add machine script as follows: add machine script = /usr/sbin/useradd -d /var/lib/nobody -g 100 -s /bin/false -M %u i have a problem with the -M option. i cannot find this option in useradd's man page. can i assume this example is incorrect?? what would be a correct example???

Hi everyone, I'm having a trouble while developing monitoring tool for queues. I'm using C# Sharp & I follow the instruction on http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+C+Sharp. My question is how can I get information about how many people are there waiting on queue with this asterisk manager? I want to make that every time there is a call come into the

Hi, [Background] I've been trying (unsuccessfully) to get a reliable RH linux distribution installed on my Intel machine with journaling on my large disks (not interested in journaling the root fs). I've tried using ReiserFS and eventually had some success, but it would seem that they are more in bed with Suse. It would appear that RedHat has chose ext3 as it's current journaling

Hi all, I'm trying to use a wrt54gl router running dd-wrt as a provisioning server for a remote installation. I've got dhcp working and I have provisioning files ready to go. I understand that I need to set bootp option 66 to point to the tftp/ftp/http server. In fact, I have this working completely with the ISC dhcp server The problem is that I don't know how to get

Hi, We just upgraded from 1.2.7 to 1.2.12.1. Everything is fine, except that asterisk seems to just crash at random. Often I can make it crash by using the ChanSpy function (which we use to monitor agents). Sometimes it will just crash on its own. The reason we were initially running 1.2.7 was because of the stability it gave us (weeks without a restart). We upgraded to 1.2.12.1 because it

Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voil? the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: "Chi lascia la via vecchia per la nuova, sa

On Fri, 2006-10-13 at 10:50 -0600, Joseph wrote: On Fri, 2006-10-13 at 07:27 +0200, Remco Barendse wrote: > > On Thu, 12 Oct 2006, Eric "ManxPower" Wieling wrote: > > > > > Matt Florell wrote: > > > > If you downgrade, let us know if it fixes things for you. > > > > > > > > It's strange that there were so many changes in the

Hi, After upgrading from: Zaptel 1.2.9.1 Asterisk 1.2.12.1 with bristuff-0.3.0-PRE-1s to Zaptel 1.2.10 Asterisk 1.2.13 with brustuff-0.3.0-PRE-1v I get the following error when connecting my Xlite Softphone: --- cut --- Nov 4 17:33:45 WARNING[4430]: chan_sip.c:1090 __sip_xmit: sip_xmit of 0x886df58 (len 486) to 192.168.9.9:31308 returned -1: Operation not permitted --- cut --- It seems to be

Periodically (as in sometimes several times a day and sometimes never) I get A channel.c: voided initial deadlock for '0x82*****', 10 retries! The ***** figure is different each time. When this happens an active call (in or out) is dropped. The setup is as follows: handset --SIP--> Asterisk 1.2.12.1 --IAX2--> Terminating supplier (don't know which software they are running).

Earlier this week 'refresh' releases of these two projects were put on our FTP servers, but due to some miscommunication on our end no announcements were sent out... so here they are :-) Asterisk 1.2.12.1 fixes one significant bug that was introduced after 1.2.10 but which hadn't been corrected yet when 1.2.12 was released (this bug is related to setting up transcoding paths between

Earlier this week 'refresh' releases of these two projects were put on our FTP servers, but due to some miscommunication on our end no announcements were sent out... so here they are :-) Asterisk 1.2.12.1 fixes one significant bug that was introduced after 1.2.10 but which hadn't been corrected yet when 1.2.12 was released (this bug is related to setting up transcoding paths between

Hi Oleg, On 01/09/14 18:46, Oleg Ranevskyy wrote: > Hi Duncan, > > I looked through the IEEE standard and here is what I found: > > *6.2 Operations with NaNs* > /"For an operation with quiet NaN inputs, other than maximum and minimum > operations, if a floating-point result is to be delivered the result shall be a > quiet NaN which should be one of the input

Hello all! I have a few problems with Snom 320 phones: Problem A - Transfer out of Queues: We have a call center with some Snoms. We are using Queue and AgentCallbackLogin. As we run * 1.2.7.1 an agent could transfer a call out of the queue using the hold and transfer buttons on the Snom. This might have been the wrong way to do it all the time I found out later, but it worked. Now we upgraded

Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. This has been working fine months but since a few days asterisk seems to die and I am not able to restart it again, I keep getting a socket in use message. This is on Asterisk 1.2.12.1, Zaptel 1.2.9.1 and Libpri 1.2.3 This is a snippet

As far as I know, LLVM does not try very hard to guarantee constant folded NaN payloads that match exactly what the target would generate. —Owen > On Sep 16, 2014, at 10:30 AM, Oleg Ranevskyy <llvm.mail.list at gmail.com> wrote: > > Hi Duncan, > > I reread everything we've discussed so far and would like to pay closer attention to the the ARM's FPSCR register

Dear list I'm testing a predictor and I produced nice performance plots with ROCR package utilizing the 3 standard command pred <- prediction(predictions, labels) perf <- performance(pred, measure = "tpr", x.measure = "fpr") plot(perf, col=rainbow(10)) The pred object and the perfo object are S4 with the following slots An object of class "performance"