similar to: extension problems

Hi Folks, I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server. When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below. *CLI> sip show peers Name/username Host

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

Hrm, All of a sudden for some reason Wait() and Playback() are returning non-zero and its causing calls on my inbound SIP leg not to complete. I'm not sure why -- Executing Answer("SIP/2181-4518", "") in new stack -- Executing Playback("SIP/2181-4518", "silence/1") in new stack -- Playing 'silence/1' (language 'en') == Spawn

I can call out perfectly fine, but when I call my Asterisk box, I don't get any sound. I see the incoming call perfectly fine and the client I am using sees it, but when I pick it up neither side can hear each other. Any ideas were to begin looking?

Hi, Using lme from the package nlme 3.1-103, I meet a strange warning. I am trying to compare to models with: library(nlme) lmez6=lme(lepus~vulpes,random=~1|troncon/an,data=ika_z6_test) lmez60=lme(lepus~1,random=~1|troncon/an,data=ika_z6_test) Both have the same number of observations and groups: lmez6 Linear mixed-effects model fit by REML Data: ika_z6_test Log-restricted-likelihood:

I figure some one here may find this interesting. I just begun work on allowing a smb home directory to be automounted upon login. -------------- next part -------------- A non-text attachment was scrubbed... Name: pam_exec.c.diff Type: text/x-patch Size: 213 bytes Desc: not available Url : http://lists.freebsd.org/pipermail/freebsd-security/attachments/20070519/19e6bd01/pam_exec.c.bin

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

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 list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Greetings and salutations to all... I'm having a bit of a problem getting a SIP phone (Xten) to call an H323 Cisco ATA-186. Both devices can call into the * and get the demo, voicemail, etc... I'm pretty sure my problem is in my configs as it feels like a stupid error and to prove this to myself I set tcpdump on the * box to capture all UDP traffic going to and from the ATA-186. If I

Hi All We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and

ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf -------- [general] port =

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

I am running a fresh copy of Mandrake 8.1, and got the latest wine source from CVS. The problem is that when I run ./configure, it says that it is unable to find my xpm development files. I have verified that they exist, and have tried modifying config as suggested on the wine help page to no avail. My xpm.h file is in: /use/X11R6/include/X11/xpm.h I will be happy to provide any additional

Guys, Ok - nowhere near as complex as most of the discussions on here ( ex telco engr for 18 years here).. But thought I'd ask for some assistance. Have just set up my first * Pbx - having a play with it and a couple of Cisco 7960 (configured as SIP) phones. The phones are tftp'ing into the server ok, and picking up the configs all ok. Everything _seems_ to be working, but I

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

Currently we have problems with our asterisk server connect with an Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 on RedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL. The Diva server is running in version 7.5 Can someone help us in reolving the errors with incoming calls? When we try to call to an extension we get these messages in the CLI: -- creating pipe for