search for: ecomtel

I have created a call file which has been moved into the outgoing directory. However the log file displays the following message: Unable to open /var/spool/asterisk/outgoing/1.call: Permission denied, deleting I have executed chmod 777 1.call on the file prior to moving it to the outgoing directory but is there something else I need to do before the file can be used by Asterisk? Any help

...an think of are: - There's a config file, though I've seen no mention of it - The actual binary shared library is modified - The system contacts Digium every time you start asterisk In the last case nothing is changed at all and I'm fine. Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd

...any confirmed details on the mailing list about it. Also, the email seems to imply that the TE405P will be fine, though it doesn't say that explicitly. Basically, is anyone using a 4 port E1 card successfully on an Intel® E7221 Chipset or similar? Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd

I'm using X-Pro connected to an asterisk server (CVS-HEAD-01/27/05-23:17:07) and after about 15 minutes in a call I get a lot of noise in my end. I don't think the other part of the call hears it. After some 10 seconds or so everything is fine again. In my CLI I get NOTICE[32322]: RTP Transmission error to 85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN

Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1

...ou have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option that allows them to dial a normal number, to also provide a normal dialtone. Should I just record one and use Background()? Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd

hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v

Hi all, I'm researching random call drops on our Asterisk and would like to make sure whether it's something wrong with our VoIP provider or with the Asterisk. I sniffed traffic between Asterisk and our VoIP provider's SIP gateway, and observed that in the middle of the conversation an RTP stream originating from Asterisk gets an ICMP port unreachable from provider's SIP gateway

I know this is a really stupid question, but I just have to ask... Where would I start if I wanted to try and develop my own firmware for a particular phone. Namely, I want to try and 're-write' the SIP firmware for Cisco 7940's. Any ideas? -Chris PS: [* put on flame suit *] why won't any of the phone manufacturer's just open-source the firmware for their phones? [*

...stion, has anyone had ohphone work for them and if so, what versions did they use at each end. Are any known compiler issues (g++ 3.0.4, gcc 2.95.4)? Are there any clients (other than ohphone) which one can use with asterisk to test it out? Thanks in advance, -- Martijn van Oosterhout IT Manager Ecomtel

Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part of a dialled number is the country code and city code ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050303/facb2a2d/attachment.htm

Hi all, We have been fighting with telco for a entire week. Today they came here with a LITE3000 to analyze what is going on. When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in

Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. I do not have a good knowledge of Asterisk, I'm not very familiar with its configuration and I've a

Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. I have mysql and mysql-devel packages installed I think!? pbx01:/usr/src/asterisk-addons# dpkg -l mysql-server libmysqlclient*dev Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed |/

Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. My musiconhold.conf is ; Music on hold class definitions ; [classes] [moh_files] default = >/var/lib/asterisk/moh-native ;default =>

...like to share? Thanks in advance, [1] http://lists.digium.com/pipermail/asterisk-users/2004-September/062882.html [2] http://lists.digium.com/pipermail/asterisk-users/2004-June/052198.html [3] http://lists.digium.com/pipermail/asterisk-users/2004-September/062872.html -- Martijn van Oosterhout Ecomtel Pty Ltd

Hi all i need urgent help our entire switchboard is down only 5 days after it came up. this is the second time this has happened and i am thinking that asterisk is not worth the trouble it gives. mostly it runs without hassle. but around 2 weeks ago during the test phase we rebooted the machine and did the normal modprobes and this error popped up. coming back to work after the weekend the

Hello, I am trying to configure asterisk 1.0.7pre to get incoming calls from an ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is that the context is not recognised in the /etc/asterisk/capi.conf I have in /etc/asterisk/capi.conf 's section "[interfaces]" the following directive context=isdn and the following directive in /etc/asterisk/extensions.conf in

Hi, I can't compile Asterisk on a Debian machine. gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o