similar to: Newbie: registration failure (fwd)

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "verbose 10" setting shows [Mar

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

Hi Hi Is there any accessible ocumentation, ie plain text or html, how to configure Asterisk. The book 'Asterisk: The Future of Telephony'' is availablly only as and pdf document and is thus unreadable for a blind user. Any pointers welcome. You can still escape from the Gates of hell: Use Linux! -- arimo

Hi All, I'm working with regcontext and sip users/peers. In the wiki, the example shows you can put this parameter in the [sipuser] context, like so: [general] lots of general parameters [sipuser] regcontext=siptest regexten=1234 Now this does not create the Noop exten priority 1 in the dial plan when the sip user registers. Now if I put regcontext in the [general] section, the sip user

Hello, I am searching for a DTMF issue on my setup ( 2 years and counting ), and I am wondering why rtp.c has code to mute DTMF ( the rtp->dtmfmute variable ), but this same mechanism does not exist in dahdi. I am sending a DTMF over SIP w/ RTP & RFC2833 to the asterisk box with the dahdi card. The dahdi card sends it out on the PRI line. Trouble is, the DTMF is echoed back and the

nautiluz wrote: >I am starting project to develop open implementation of some (naturally open >codec) for simply designed embedded devices which can be used by small to >big VoIP operators or hobbyists which wants to build their own small and >low cost VoIP phone. >Speex seems to be great choice. And greater will be if there should be >some guys who want to help ;-). Sorry

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients(using linphonec). In a proper context, I have mentioned extensions 107 as simputer@X.X.X.X (x.x.x.x=asterisk server ip) Asterisk Sever-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:110@siptest.dmclub.net) B is connected to the Asterisk server via VPN B is registered (and has successful bi-directional conversations with other users on the

Is there a linux sip or iax phone that will autoanswer and connect to the console or soundcard? I found linphonec but it does not autoanswer from what I can tell. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050916/8bc76bbb/attachment.htm

Hi, I've got this PXA270 board set-up with Linphone 1.2.0 and am trying to get linphonec to work with Asterisk. I have the echo test working, but when I dial in to this, to voicemail or anything else using Playback() to play a sample, I hear nothing for ages (10-15 secs) and then little sections. With the echo test, I get the tail of the message (...pressing the pound

Hello I have changed type 'caldav' to 'ical', but still no succes : [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:117 auth_credentials: Invalid username or password for iCalendar 'cal1' [Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:150 fetch_icalendar: Unable to retrieve iCalendar 'cal1' from

All, I made some progress, but it seems the further I go with clustering the harder things get. Hmmm, I guess if it were easy, it would be documented...... Anyhow, I have 1 * server as the DUNDi peering master with a ttl=1. The only function of this server is to lookup where other sip peers are registered and forward that info on to the requesting * server. I have 4 * servers accepting

Hi, I am trying to setup my own PXE boot server. I tried several PXE bootable Linux-Distributions. For example if I use the ubuntu netboot image from [1] it works quit well but there are a few other images they do not work in my case e.g.: RIPLinuX [2]. A friend tested this image on his PXE boot server with success. I also checked the download with md5sum. In my case I can see the boot menu [3].

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Hi all! I am using samba server at the company i work. Samba version is: 2:3.2.5-4lenny9 Some settings: os level = 255 domain logons = no wins support = yes domain master = yes local master = yes preferred master = yes i have dhcp configured witch sends wins server address (this samba servers address) to all clients. After I (re)start samba within a minute or so all computer of our network

sieve deliver still doesn't build with Sun cc. any suggestions ? thank you. cc -DHAVE_CONFIG_H -I. -I. -I../.. -I/export/home/sho/src/dovecot/rc8/dovecot-1.0.rc8 -I/export/home/sho/src/dovecot/rc8/dovecot-1.0.rc8/src/lib -I../../src -I/opt/SUNWconn/crypto/include -xjobs=4 -c comparator.c -KPIC -DPIC -o .libs/comparator.o "comparator.c", line 149: syntax error before or at:

cc -DHAVE_CONFIG_H -I. -I. -I../.. -I/opt/SUNWconn/crypto/include -xjobs=4 -c file-set-size.c "/usr/include/sys/resource.h", line 126: incomplete struct/union/enum timeval: ru_utime "file-set-size.c", line 26: warning: implicit function declaration: ftruncate "file-set-size.c", line 38: warning: implicit function declaration: pwrite cc: acomp failed for

We have a local time server and all of our machines are pointed at it for the time. How can the clock drift by a day and a half? [root at devserver21 ~]# date Fri Aug 13 14:43:29 EDT 2010 [root at devserver21 ~]# rdate -s 192.168.1.67 [root at devserver21 ~]# date Thu Aug 12 07:02:39 EDT 2010 [root at devserver21 ~]# cat /etc/ntp.conf | grep -v ^# | grep -v ^$ restrict default nomodify notrap