similar to: Zap gives no ring to the caller...

Somone in europe have had succes getting Callir ID showed on a phone screen conected to an Handytone 286 ? Adri? Vidal -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 235 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/e5514052/attachment.bin

Hi Everyone, I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones occasionally go into "No Service" mode. The POE switch doesn't seem to be the problem as it's tested fine. I think the router sometimes gives up and comes back quickly. Or something of that nature. However, the connections are maintained if a call is going on because there are peer to peer

Hello! I'm having a bit of an issue with one of my installations that I cannot figure out. For some reason, when two people are in a call (both local to the * box, same subnet, pure SIP), the call will randomly be placed on hold and provide MOH to the other party. We're using Polycom IP430 handsets almost exclusively for this installation. Can anyone think of a reason why a call would

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

Someone have info about install an AVM fritz into FC3 ? I'm getting problems with kernelcapi, after succesfully installed the fcpci support. Thanks -- Adri? Vidal adriavidal@gmail.com Mail is better with 1Gmail

Someone can confirm the new release is out? Haven's seen any post about it!!!!! -- Adri? Vidal adriavidal@gmail.com

Hi, does the hint extension work together with the Snom phones in stable? I don't get an error in the dialplan, but it does not work either. On SIP/26 I want to monitor SIP/22. This is what I do right now: extension.conf [incoming] exten => 955,hint,SIP/22 exten => 955,1,Dial(SIP/22) sip.conf [26] ... subscribecontext=incoming ... Running Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b TIA

I'm getting "Got SIP response 406 "Not Acceptable" back from 10.0.1.2" occasionally when try to dial to SPA942 , anyone has any idea on this before i consider Firmware upgrade? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080326/2fd2c557/attachment.htm

Hi list, I have for a year I have an account to call with broadvoice from about 3 days beginning a not registered problem of, asterisk shows to a message of error with the DNS, and my dns this working fine WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for registration to 908XXXXXXX at sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds) [Dec 16

My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the interface it has a place for me to record new IVR menus. I have to dial *77 to begin recording

Have someone running fine Alert-Info with a Snom 370 ( System Information: Phone Type: snom370-SIP MAC-Address: 0004132661BD IP-Address: 192.168.10.170 Firmware-Version: snom370-SIP 7.3.14 14961) i've tried exten => 200,1,SIPAddHeader(Alert-Info:<http://www.notused.com>\;info=alert-external) exten => 200,n,Dial(SIP/${EXTEN},30) Can see into the phone SIP trace is

Hi everyone, I have been long working on a project (http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the

I'm testing some of the new features of Asterisk 1.8, but seems an impossible mission to make the calendars run. cli show empty: **CLI> calendar show calendars Calendar Type Status -------- ---- ------ * And i can't see anything into the log, calendar.conf is ignored. Any sugestion? -- -- Adri? Vidal -------------- next part -------------- An HTML

Hello everybody, I'm having trouble with samba 3.5.11 and an access 97 database. I work in a business with 5 users that use a mdb database over samba, and for the first to make a query everything works fine, but if some other user tries to make a query when the first one is doing it, the database works really slow, like if the .mdb file is being blocked (that's what I suppose). I've

Hi have what might be some elementary questions. Really, what I"d love would be for someone who has had good success to publish his/her configuration files and, maybe, the output from ifconfig. At this point, when I see the not-so-good performance I"m getting, I don't realistically know if I'm in the right ballfield. It seems to me, that with so many mellanox cards out

The Asterisk.org development team has released four new versions of Asterisk to address critical security vulnerabilities. AST-2008-002 details two buffer overflows that were discovered in RTP codec payload type handling. * http://downloads.digium.com/pub/security/AST-2008-002.pdf * All users of SIP in Asterisk 1.4 and 1.6 are affected. AST-2008-003 details a vulnerability which allows an

The Asterisk.org development team has released four new versions of Asterisk to address critical security vulnerabilities. AST-2008-002 details two buffer overflows that were discovered in RTP codec payload type handling. * http://downloads.digium.com/pub/security/AST-2008-002.pdf * All users of SIP in Asterisk 1.4 and 1.6 are affected. AST-2008-003 details a vulnerability which allows an

I've just got a Snom 190 phone with which I'm really pleased. I can get the LEDs on the keys to light in response to an extension being in use which is cool, but there's a feature I'd like to implement. I'd like to light one of the LEDs on the keys to light if I have night-service enabled. My Night service is currently controlled by a variable in the database. I could go

I have added a sip user in sip.conf. user name is 819,context is c819. and I have added the follows rows in extension.conf. like [c819] exten => 1,1,Answer exten => 1,2,SetVal(voicemail=${exten}) exten => 1,3,Dial(SIP/${voicemail}) It appear a error message(pbx.c:1279 pbx_extension_helper: No application 'SetVal' for extension (c819, 1, 2)) when i dial 1 from 819. The