similar to: zapras - and kernel ??

hi! I'm trying to get zapras working in GSM csd network. Whenever a dialup call is initiated from the mobile to the * gateway the following appears in the log and zapras terminates. Phone gives the error dialup not answered. ==> /var/log/messages <== pppd[2310]: Plugin zaptel.so loaded. pppd[2310]: Zaptel Plugin Initialized pppd[2310]: Using zaptel device 'stdin' pppd[2310]:

hi everybody, i'm tryint to install a asterisk system which acts as a dialin server using a Digium Wildcard 205P. acording to http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS i need a patched version of pppd, but it does not compile on my system. Linux box 2.6.17-gentoo-r8 #1 SMP Tue Sep 26 13:17:23 CEST 2006 x86_64 AMD Athlon(tm) 64 Processor 3200+ GNU/Linux gcc -4.1.1, glibc-2.4

Hi, I am trying to use Asterisk cmd ZapRAS (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ZapRAS), I pathed the ppp daemon ftp://ftp.digium.com/pub/zaptel/misc/, but when I try to use it, I obtain the following error: Connected to Asterisk 1.2.4 currently running on TSU-R1 (pid = 7242) Verbosity was 0 and is now 44 -- Accepting call from '123456789' to

I'm trying to use ZapRAS to enable ppp connection through my E1. After the ZapRAS command is executed, all sound is crappy on all lines! The only solution is to reboot the machine (or halt it, and then power it on since Digium's hardware doesn't like reboots). Anyone know how this can happen?! I'm using * 1.0.6 on Dell PowerEdge 1850 which are told (too late though) not to work

Hi, I'm trying to get ZapRAS working but not getting very far.. Asterisk CLI shows: WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No child processes and /var/log/messages shows: using the plugin option requires root privilege Can anyone shed any light on this and any fix? Googling the error doesn't find much.. I'm not sure what 'plugin' it is talking about,

hi all Where can I find documentation about how to setup ZapRAS? What I want to do (optimally) is to allow for automatic dial-up to external sites, each having an ISDN router. Today we use a small ISDN router for this, but it'd be a lot better, IMHO, to have asterisk do this (functioning as a ISDN router), as we may cancel our BRIs then. Is this possible? And if so, how can I do it? I

Hi I try to use zapras. I am using zaptel-bri-0.0.2 I compiled and patched pppd from ftp://ftp.digium.com/pub/zaptel/misc/ pppd is /usr/sbin/pppd Any idea whats going wrong here? Thomas -- Accepting call from '95' to '8526' on channel 1, span 1 -- AGI Script nuller.agi completed, returning 0 -- Executing ZapRAS("Zap/1-1",

[ZapRAS triggering a kernel panic] >> Kernel panic: Aiee, killing interrupt handler! >> In interrupt handler - not syncing >> HDLC Receiver overrun on Channel Tor2/0/2/25 (master: Tor2/0/2/25) > > Hrm, I haven't seen this before. Please contact me off-list and I'll > give you more debugging instructions that may be helpful, as well as > enquire additionally

Hi! I've been trying to get ZapRAS or PPPD to work. Neither does! All i get is LCP: timeout sending Config-Requests But after trying, all voicelines get crazy! It sounds like robots when somebody calls! And since the zaptel drivers can't unload (the server hangs totaly if I try!), I have to reboot the whole server! The robot-voice is only on our side, it sounds fine at the other end.

I have an issue where * tries to route a call over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip

I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List I'm trying to create a conference room using H323 channels. If i start asterisk normally (service asterisk restart) and connect to cli using -vvvvvvvr options, when a user enters the Conference, asterisk says "You are the only ..." and then dies, withou any error message, nothing at all. But, if i start asterisk with cli

Hi All, I'm on a middle of an asterisk installation/configuration for my company and I'm testing the TLS configuration. For this reason, I used the ast_tls_cert script to build the ssl certificates for my server. On sip.conf file: tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 and on

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Hi, I had compiled support for MYSQL_FRIENDS and it works for SIP, but when use tiwh IAX2 I have some problem, I can register with a client, but when I try to make a call I got this error: Jul 17 12:52:03 NOTICE[229387]: chan_iax2.c:5183 socket_read: Rejected connect attempt from <IP-ADRRESS> When I google'ed this problem I can see other users also found this error (bug ?) But no-one

I am having trouble getting Asterisk to compile the zaptel stuff. Here are the specifics: Linux Kernel 2.5.9-42.0.8.EL Asterisk 1.4.0 I compiled libpri, zaptel, asterisk and asterisk-addons (in that order). This is a fresh install of CentOS. Following the CentOS install, I did "yum -y update" until there were no updates left. Here is my src directory: drwxr-xr-x 24 root root

Hi list, something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both having an extension [115], one as type peer (caller side 1.4) and one as friend (callee side 1.8). Phones from both location connect to Asterisk from LAN. Router are Linux boxes. Connection between the 2 sites is done like this: On the callee side [115] ;callee type=friend host=dynamic secret=otherSecret

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP