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...he Originate Action (using AMI), what I need to put in the context > parameter? The GoSub will jump to a special context. > Thanks again.... > > > > > ------------------------------------------------------------------------ > Date: Tue, 15 Sep 2009 17:17:38 -0500 > From: mmolina at millenium.com.co > To: asterisk-dev at lists.digium.com > Subject: Re: [asterisk-dev] MeetMe in Macro > > Anahi Ludue?a escribi?: > > Hi people, I have the following context in my dialplan: > > [macro-meetme-part] > exten => s,1,Set(PART_OPT=${ARG1}) &gt...

Hi everyone, I was just wondering, does anyone managing production asterisk servers migrated successfully from 1.4.2X to 1.6.0.X? I would like to see if there are some successful cases. Is your 1.6.0.X behaving well, with acceptable stability? Please share your experiences. Thanks, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57

Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi

Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here: http://ubuntuforums.org/showthread.php?t=254623 Should i try in the pass this parameter in the

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

I think there is a problem. If we are coding PCM 16 mono, then, when we decode the samples we should obtain PCM 16 mono again, but wave data over 16bits is extracted from the decoder. If the input level is reduced we could loss wave information. Thanks for your help <p>Jean-Marc Valin wrote: >> we are using speex 1.0.3 under windows. We code samples with

Hi all, I'm experimenting with the MySQL realtime logging of queue log, using backports for 1.4. When I configure the logging of queue log as realtime, the previous way to save it onto the plain text file stops working. Is there a way to make it work both ways, just like the CDR works, where you can have several backends and still save the CDR on a plain text file? This could be useful

Hi all, I just upgraded a production server to asterisk 1.4.25, compiling with the following: [*] 1. DONT_OPTIMIZE [*] 2. DEBUG_CHANNEL_LOCKS [*] 3. DEBUG_THREADS [*] 4. DEBUG_FD_LEAKS [ ] 5. LOW_MEMORY [*] 6.

Hi all, I'm testing the persistent dynamic queue members functionality on 1.6.0.10. The queue members are agents defined in the agents.conf file. When I issue an asterisk restart and check the queue members again on the CLI, all of them are listed as /invalid/ and there is no way to change this but to unload app_queue.so and load it again. My guess is that the internal AstDB queue

Hi folks, I have a simple question regarding attended transfers. I have some queues where agents take calls and I have configured attended transfers between queues. That is, the agent dials the attended transfer extension that routes it to the aproppiate transfer queue where the second agent answers and they both talk for a while. Finally the transferrer leaves the call with *, connecting

Hi folks, I am struggling to install cdr_pgsql in asterisk 1.6.0.26. When I do the ./configure, it complains about the function PQescapeStringConn not existing in -lpq, so when I do a make menuconfig, I can't select the cdr_pgsql module. I am using CentOS 5.4 with the yum PGDG repository for 8.4 version. Some of the installed packages are postgresql-libs, compat-postgresql-libs,

Hello list, I don't know if anybody faced this issue, but I finally found a workaround. I am using an external program with an AMI connection to originate outbound calls to Local/ channels, and on the dialplan context I dial outside with the corresponding trunk according to the prefix of the dialed number. The problem here was CDR, more precisely the uniqueid. Reading these mail lists I

Hello, we are using speex 1.0.3 under windows. We code samples with values from -32.000 to 32.000 (signed short), but when we decode these samples the values range from -45.000 to 45.000. What is the problem? <p> Thanks --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message

Hi to all, I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free ()

Hi everyone, Does someone know why the solution for bug 13115 (https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=13115) was made only for trunk? Having that this bug went solved more than a year ago, it means that all the 1.6.X.X branches have it applied already? Can this be backported to the 1.4 branch? This could be another good reason to upgrade to 1.6.0.16 after I do some

Hi all, Anyone know why this happens? Mem: 524288k total, 508120k used, 16168k free, 0k buffers Swap: 0k total, 0k used, 0k free, 0k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1 root 15 0 2152 664 576 S 0.0 0.1 0:49.26 init 7398 root 18 0 10172 2904 2312 S 0.0 0.6 0:00.21 sshd 9856