similar to: Inserting a wait in a sip dial

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! I have a working asterisk-setup with four sip-clients. Everything works great but when the users call someone the phonenumber shows up on the receiving ends callerid-display. To correct this my provider told me to send #31# before the phonenumber, tried this with: Dial(SIP/#31#${EXTEN}@provider) but my asterisk tells me that it isn't a valid extension. The INVITE looks fine,

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 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

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

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! I'm trying to read individual files from a ZIP archive, using the unz() function. Some of the files contain non-ASCII characters and I'd like to avoid unpacking them in a temporary directory. My problem is that unz() seems to ignore the encoding="latin1" option I need to read the non-ASCII characters properly. I can't find a clear indication in the documentation that

Is anyone interested in some part of this filesystem to figure out how it failed? Or can I erase and start again? Kind regards, -Evert Vorster- On Sun, Aug 8, 2010 at 8:18 AM, Evert Vorster <evorster@gmail.com> wrote: > Hi there. > > I have a btrfs on a raw device. (/dev/sda insted of in a partition, like > /dev/sda1 ) > The device in question is a USB hard drive with a 1TB

>From my extensions.conf: exten => 888,1,Answer() exten => 888,n,WaitExten(20|m) exten => 888,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) This should: * answer * wait 20 seconds for an extension with music on the background * pass the call to that extension on ${SERADDRESS} What am I doing wrong here? I don't even get the background music while WaitExten is active. I doubt that

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 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! 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 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! 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! 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