similar to: choppy sound after 15 minutes in a call

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

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any

I'm asking because I'm planning to install multiple machines from the same image and I need to know what file(s) I need to backup/restore to make sure I don't lose my licences in the process. The only options I can 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

Hi, I know this is slightly off topic but I figured the knowlege here is probably the best on the subject.. I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box.. These phone will be behind an ADSL router using NAT... I don't want to setup another Asterisk system in each office so IAX is not an option.. I could use

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! I'm trying to set up the possibility for users to call my Asterisk from the net. The Asterisk is behind a Intertex IX66 in which a have set "Static domains" so it forward all calls for my hostname and external IP to the * box. when somenone calls at lars@<external-ip> it all works, but if they call lars@<hostnmame> it wont work. At my Asterisk console I get the

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

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? [*

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, 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 =>

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, Is this card the same as the T410P, after all, it's made by Digium. There's one prior reference on the mailint list[1] but it didn't answer the question. There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Any info anyone would like to

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Hi, I'm going to be setting up some machines with 4 port E1 cards from Digium and I'm being told that TE410 is incompatable with several Intel chipsets including the ones in a lot of Dell server systems. Is this true? I can't find 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.

Hi, I see applications for signalling busy, congested, ringing, progress etc, which I understand can be provided either in or out of band. But all I want to do is generate a dialtone. This obviously can only be done in band. There is code for generating the tones when you have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option