similar to: Cisco7905 keeps forwarding to voicemail

Hello, Does anyone know how to set the outgoing CallerID properly when using Snom200/SIP/CAPI/BRI? Following doesn?t work: exten => _0.,1,SetCallerID,526910 exten => _0.,2,Dial,CAPI/526980:${EXTEN:1} Asterisk writes: *CLI> -- Executing SetCallerID("SIP/226-ada0", "526910") in new stack -- Executing Dial("SIP/226-ada0",

Hallo all, I am trying to get * with chan_capi and a ptp-ISDN with 4 lines on a AVM C4 card to work. But weather inbound nor outbound is working :( My capi.conf: ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] mode=immediate isdnmode=ptp msn=8993 incomingmsn=* mode=immediate controller=1,2,3,4 softdtmf=1 ;accountcode= context=demo

O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps telling me while startup: [chan_capi.so] => (Common ISDN API for Asterisk) Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr1 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr2 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr3 Jan 24 14:30:47

Hi, i use chan_capi 0.3.1 with asterisk (stable branch cvs) and 3 x c4 active ISDN card. From Controller 1 - 7 there are no problems making calls between asterisk and the pstn. But when i make calls from controller 8 - 12 i get on every controller (8 - 12) a segmentation fault in asterisk :( I tried different linux distributions (gentoo 2004.1, redhat 9.0 , suse 9.1) but same error.

Good day, does anyone have AstTAPI running for incoming calls, and would like to show some examples. My setting right now looks like this: sip.conf -------- [22] type=friend dtmfmode=info username=22 mailbox=22 secret=privat host=dynamic context=privat canreinvite=yes callgroup=1 incominglimit=2 extension.conf -------------- exten => 123,1,noop ;Hint(SIP/22) exten =>

WARNING[<pid>]: CAPI[contr3/123456]/178 already has PBX structure?? WARNING[<pid>]: CAPI[contr3/123456]/178 already has a call record?? WARNING[<pid>]: CDR on channel 'CAPI[contr3/12345]/177' already started WARNING[<pid>]: Thread 1109916592 Blocking 'CAPI[contr3/123456]/178', already blocked by thread 1116277680 in procedure ast_waitfor_nandfds

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Hello, I'm looking for any comments or user experiences from anyone who is using 7912G phones with SIP. Any installation issues? Usability problems? Do the features seem to work, etc...In short, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation

I have two Sipura 3000s, one for our main phone line the other for our fax line. I think I need to handle each device in seperate context sections. Both contexts use the s extension and things are not working as it was before I added the second Sipura for the fax line and additional context. Is it a problem to have two contexts with s extensions? What is the proper way to handle this senario?

I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. /carmi

Hello All! I am in the process of assembling an asterisk-based phone system for my office - using SPA-3000s to connect the network to the PSTN. I am wondering if anybody else can get (or has already seen) the same behaviour out of their 3000. The short version: Send multiple Calls to the SPA's FXO port at the same time it is re-registering with Asterisk. SPA HTTP Configuration:

Does anyone have any experience with the Audiocodes MP-108 FXO gateway? I'm looking to get one for incoming PSTN lines. In particular, does it pass caller ID information to Asterisk? I currently have a Mediatrix 1204 but Caller ID does not work, even though the specs say it does. All it sends are the names of the ports set up internally on the gateway (ie. "pstnline1" etc) when

Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn't work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL:

According to the mongrel_cluster.yml file you provided, you''re starting only 14 mongrels, but in your proxy balancer config you have 30 mongrels listed on ports 21000 - 21029. In this scenario, if apache tries to proxy to any ports higher than 21014 then you''ll get a proxy error as a mongrel doesn''t exist on that port. Or am I missing something? -----Original

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

I''ve posted this to rails-deployment as well. I have to administer a medium size rails app (1''5 million requests each day), recently I''ve switched from lighttpd + fcgi to apache + mongrel. In the following lines I am going to describe the platform: All machines are running Debian Etch, with 4 gb ram and dual core intel32 processors. Web server runs debian''s

Hi. Am new to this concept but have been requested to add VOIP capability to a small office phone system. They currently have 4 standard analog lines running into a PBX feeding 16 phones, with all the usual features, call transfer call hold internal calls etc. would the following seem reasonable ? asterisk server:- ( what specs ) cat5 > broadband (VOIP) 4 FXO's for incoming PSTN

Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Thank you all . --

A problem here getting winbind traffic to be encrypted using Kerberos. I have set up a test environment with a pair of servers (actually lxc containers): - samba server (ubuntu 16.04, stock samba 4.3.11) - client machine (ubuntu 16.04) joined with "net ads join" and winbind The client machine has the following in /etc/samba/smb.conf: ------- [global] #netbios name = client-ad

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only