similar to: Wrong ID going out...

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert

Hi all! I need a simple plan for the following: *answer call *wait for 4 digit extension *send call to 4-digit extension entered. I tried the following, but that doesn't work... exten => 998,1,Answer() exten => 998,2,Background(agent-newlocation) exten => 998,n,WaitExten(20) exten => 998,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) WaitExten obviously does not fill EXTEN with

Hi everyone! situation: Asterisk-server A: 192.168.11.6 Asterisk-server B: 192.168.2.44 server B contains a register => username:password@192.168.11.6 But... when I boot it, I get: Registered to '192.168.11.6', who sees us as 10.138.3.2:4569 Why doesn't server A see server B as 192.168.2.44?? All other traffic going over these lines has no problems with this. The

Hi everyone! I'd like to create the following: a user picks up the phone (gets a dial tone), dials '0' for an 'outside' line, gets a second (different?) dialtone, and is able to enter an external phone number. How do I implement this in extensions.conf...? Regards, Evert

Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf ***** [ip-incoming]

Hi All, After searching and didnt found it, im just sending my situation here, maybe someone knows where i should look. Im using Asterisk 1.6.1.10 Internally the user with a sip phone dials a number for instance 0623456789 It goes fine to the specific dial rule: which is: exten => _0[6].,2,Dial(SIP/0${EXTEN:1}#@xs4all-out,60,tTwWkK) This works fine without a charm, but the situation is that

Hi all! We use a PHP-portal for management of our projects & contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so

Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 11111 -> ext. 1 913 - 22222 -> ext. 2 913-11111 & 913-22222 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the

Hi all! Working on the following brain-scratcher. I am setting up a Trixbox system for someone who uses 'provider A'. Everything works fine, except for the IVR: keypresses by callers are not being detected. Just for testing I added my own provider, 'provider B' to their system. And then the IVR works! Is there any possibility that the config on the provider-side is causing this

Hi all! Am I the only one with this problem? I doubt it... The problem is that I have a couple of shares of a W2K server mounted with Samba on my (Gentoo) Linux. This works fine, until the W2K server gets rebooted. After that the shares are just timing out, and they are impossible to unmount/remount... :-/ How do I prevent/fix this problem? Regards, Evert

Hi all, I have been looking at various alternatives for a system that runs several Linux & Windows guests. So far my favorite choice would be OpenSolaris+ZFS+RAIDZ+VirtualBox. Is this combo ready to be a host for Linux & Windows guests? Or is it not 100% stable (yet)? Greetings, Evert This message posted from opensolaris.org

Hi all! Does NUT offer any support for the GE Digital Energy NetPro 19" UPS? I have one of those beasts here, connected to a Linux machine via it's own 'serial' cable. Regards, Evert

Hi everyone! Who can tell me how I can access my voicemail? When I dial the voicemail on my Cisco 7960 I get access, but when trying to enter my mailbox number it seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem perhaps? Any suggestions on how/where to fix this...? Regards, Evert

Hi! If I install a CAPI-compatible ISDN-card in my server, will that: a) enable me to connect that server to the public phone net b) allow me to connect an ISDN phone to the server and use it as a SIP-phone c) all of the above? Regards, Evert

Browsing in the 2003-September archives, I found out how to start dovecot from inetd (using imap-login instead of imap). Still I've got a few problems. I prefer not using inetd, but that seams impossible (see my 1st email). And telnetting to port 143 results in an immediate connection close. Hmmm, I'll continue exploring the archives. Regards Evert

Hi everyone! I think I stumbled over a bug in the latest R 2.9.2 patched for OS X: > R version 2.9.2 Patched (2009-09-24 r49861) > i386-apple-darwin9.8.0 When I try to sort latin1-encoded character vectors, R sometimes crashes with a segmentation fault. I'm running OS X 10.5.8 and have observed this behaviour both with the i386 and x86_64 builds, in the R.app GUI as well as on

Hi All, Maybe someone knows this, im using dahdi in combination with a TDM400, where 2 analog PSTN lines are connected. The weird thing is tho that when someone calls the analog lines it goes perfectly fine, the line comes in and all works ok. Except: Sometimes the callerid from the caller is not the complete number, but only a few random numbers from that phonenumber, and sometimes its complete.

Hi everyone! The following: Any calls coming in on extension 12121212 should get a message telling them to dial the extension of the person they are trying to reach, and then press #. The call should then go to the entered extension. This is as far as I got... *********************************************************** exten => 12121212,1,Wait,1 exten => 12121212,2,Answer exten