similar to: Outgoing works, incoming doesn't...

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 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 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! 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 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! 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! 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

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 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

Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled

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

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

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

Hey, I just started trying to use the qualify=yes option on my Cisco 7960 SIP phones. Of the 13 I have, 2 of them seem to loose their registration with asterisk on a regular basis. I see lots of these lines: -- Registered SIP '3030' at 62.74.107.1 port 58825 expires 60 in my console. But I only see them for 2 extensions. Never see them for the other 11. All 13 phones have the exact same

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. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

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 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

When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found