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Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I

Hello, I've been doing some testing lately on Asterisk. I've had some problems with it using 100% cpu at times. One time, it held the 100% cpu usage for 12 seconds. I've tried the latest -head and -stable as of yesterday. I saw Kevin just made a commit, so I'll try updating again. I was wondering if there is a specific version that someone has done heavy testing on (handling

...y time I do this, it's great for a day or so then it acts up. Before after it started, even if I rebooted it would immediately start up. On 5.1, it is only hanging for that process and everything else is fine. I can still login, webserver responds, etc. Here is a little info: FreeBSD devel.neoninternet.net 5.1-RELEASE-p2 FreeBSD 5.1-RELEASE-p2 #0: Sat Aug 23 20:12:41 PDT 2003 kevin@devel.ph.cox.net:/usr/src/sys/i386/compile/SLURPEE i386 CPU: AMD Athlon(tm) XP 2600+ (2086.51-MHz 686-class CPU) real memory = 1073676288 (1023 MB) ad0: 117246MB <Maxtor 6Y120P0> [238216/16/63] at ata0-mas...

Hi, I used to have a Grandstream phone connected to Asterisk a few months ago. Worked just great! Then today I do a new install, rather than an upgrade, and all of a sudden I cannot check voicemail with it. No problem calling or receiving call. It simply speeds through the vm greetings but I cannot hear them. If I check the same VM with an analog phone it works fine. So I wanted to check

Hi Peoples I am having problems with meetme, in that it responds with "conf-invalid" when I dial a conference number. I notice that there is a note with regards to ztdummy, and the need for that to be loaded. Is this still the case? Is meetme dependent on this module? I do NOT use zaptel cards in my system, and there for zaptel is not loading. Can anyone shed some light

Hi. My bad, but I have not been tracking the -stable list and I just updated -stable on a production machine. It seemed to work just wonderful, as normal, for about 10 minutes then it started to deny connections and when you run commands, it would just hang. I have done a reboot and fsck everything. It is up now but things are still strange. Here's the output. I just updated -stable at

Hi. Thank you to Olle, the Wiki, and Trollphone. I found the Trollphone Rate Engine listed on the Wiki at: http://www.voip-info.org/wiki-Asterisk+addon+rate-engine I'm not familiar with LCR yet, but this is something that I need to do. I have it all installed but am not familiar with the terminology. Can someone help me maybe figure out what some of these fields are for? mysql>

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of empire underground Sent: Friday, January 09, 2004 1:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Screen Pop & Remote Agents > can I put a .csv file in the sql DB and have it dial from there? and will I be able to set a > Dial Plan to

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Hi. I've noticed a problem with the expression parsing in Asterisk. If the variable is not defined, I will get a parse error. Yeah, there are ways around it, but I would think that it should return false if 0, null, or undefined. I would change it, but I have no idea about bison and I only have very basic C skills. There was a bug opened on this, and there was a valid work-around posted,

> Hackish, but it works. I've used it myself several times. This, of course, > assumes you are using Zap channels for incomming calls. If not, then you'd > need to find another way to listen to incomming calls - perhaps ChanSpy, tho > i've not been able to get that to work - crashes my * box with CVS HEAD. > > -josiah It does this to me too. We should send in a

> I'm still having my (apparantly) timing problem, but I'll do some more > testing and make a separate thread for that. I'm generating an > outbound call through Asterisk. The inbound audio is good, but the > outbound audio is sometimes staticy. This seems to happen only at the > start of the call. These are short test calls, just playing a "weasels > ate our

Hello, I have figured out that my audio problem was just how I was converting the sound files. I am trying to convert the Asterisk gsm files to ULAW. I just did a: sox file.gsm file.ul, open it in Audacity. I used: Project, Import Raw, U-law, No endian, 1 channel, start offest 1 byte, sample rate 8000hz. The file sounds fine in Audacity. Now, if I do a record on Asterisk, using pcm, au, or

Hello, Maybe I'm missing something here. What is the proper way to use RTC with ztdummy now? I'm using -HEAD from a day or two ago on Linux 2.6.11.11. In zaptel/Makefile, I changed CFLAGS to: CFLAGS+=-I. -O4 -g -Wall -DBUILDING_TONEZONE -DUSE_RTC #-DTONEZONE_DRIVER and I get.. make -C /lib/modules/2.6.11.11/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory