similar to: sip register incoming call contexts?

Displaying 20 results from an estimated 3000 matches similar to: "sip register incoming call contexts?"

2006 Oct 17
1
looking for a cleaner way to do something
I have two numeric vectors each of length 17 and each is named the exact same way. so obsnum ppppp ppppm pppmp . dot dot dot...... temp1 is 1417 52 63 85 obsnum ppppp ppppm pppmp . dot dot dot...... temp2 is 1213 41 50 97 what i want to have is a resultant matrix with 2 rows and 16 columns where the 16
2005 Jun 02
3
CLUELESS NEWBIE needs help making an outbound sip call to PSTN
I'm going to try and ask this again and keep it short and as too the point as I can while still providing enough info to be of use. PLEASE advise if I am going about this wrong or asking too much. I'm seriously doing my BEST to throughly read the docs and try a bunch of things BEFORE coming here to ask and possibly annoy. If is documentation that explains thsi process in terms that
2003 Sep 10
9
Free World Dialup (FWD).
Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers
2008 Jul 17
1
blktap complaining what does it mean ?
Hi, I''m a newbie to Xen. I installed a DomO CentOS 5.2 and several DomU (centOS, Debian and Ubuntu). It worked very well. Yesterday I experienced a power cut. I now experienced some problems. For example, when I start a DomU with the "xm create whatever_I_want" command, I get : Using config file nnnnnnn tap tap-10-51712 : 2 getting info Started domain xxxxxx and just after,
2005 Sep 05
2
USING TWO ACCOUNTS WITH BROADVOICE
Hi, I have two accounts with broadvoice. Now, I want to be able to distinguish between them. I though that this would be simple by adding "/EXTEN" at the end of the register statement. For example: register => num1:pass@sip.broadvoice.com/1000 Unfortunately, this is not working. When I call into my box I hear busy tone. My config looks like this: [root@voip asterisk]# cat sip.conf
2006 Apr 10
7
Asterisk BRI in the USA
Hey all, It such a shame that BRI technology is such a flop in the USA. For a small office such as mine it would be a great product. So her goes my question.... What is a known asterisk working BRI card that will operate in the USA. I need to weigh price/quality. I need to do DID/DDI (or what ever you want to call it). Asterisk will do everything else I need. The ILEC has at the other
2003 Feb 22
1
SIP register= bug?
I am seeing some very peculiar things in the routines that REGISTER my * server with several accounts. I saw this on my console: . . . NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout):
2008 Jan 04
5
win32-api callback causes ruby to application error (crash).
# I originally intended to post this question here. # But it took some time for subscription authorization process. # So I have posted the same question to the forum when I was waiting # for the subscription confirmation notice. # It seems to me the forum is not so active, let me drop here too. Hello, The following WinSNMP trap receive program ''win_snmp.rb'' causes ruby
2007 Mar 29
3
Asterisk hangs up SIP call after 6 200 retransmits
I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all
2008 Sep 30
5
Corrupted transaction log file / record size too small
I recently upgradeded dovecot on one of our servers from version 1.0.10 to version 1.1.3. Ever since, we've been seeing occasional errors similar to this sequence (with the username and IP addresses elided): Sep 30 00:09:56 alcor dovecot: pop3-login: Login: [4954], XXXX, NNN.NNN.NN.NNN Sep 30 00:09:56 alcor dovecot: wrapper[5006]: pop3, XXXX, NNN.NNN.NN.NNN Sep 30 00:09:56 alcor
2005 Mar 27
6
How to use multiple VOIP provider trunks
I have been able to setup three different providers successfully, but only one at a time. I would like to have all active in a fail over configuration so that one failing would not be noticed by the users. I know it's probably easy to configure but I have not been able to find out how. Can anyone give me an example? Chris Mason
2005 Jul 04
2
Extensions will not go to voicemail
I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp("IAX2/netconcepts@nnn.nnn.nnn.nnn:4569-5", ""Dial 710"") in new stack -- Executing
2005 May 13
4
1-800 with FWD
Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"
2005 Jan 18
3
Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore)
I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the
2003 Aug 19
1
Problem with * server and FWD
I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>
2004 Mar 16
1
VMware Printing Problem - Access Denied, Unable To Connect
I see you are using cups. I had the same problem It is a cups problem. First install a RAW printer in CUPS. Second allow cups to receive jobs from a remote host. By default it doesnot. -- Groetjes/Regards Kees van Hoof
2003 Oct 18
1
'passwd chat' for Debian Woody password sync
what is the 'passwd chat' line for Samba 2.2.3a-12.3 on Debian Woody that enables password synchronisation with Windows 2000 clients in a domain? I currently have this in smb.conf but it doesn't work: unix password sync = yes passwd program = /usr/bin/passwd %u passwd chat = *Enter\snew\sUNIX\spassword:* %n\n *Retype\snew\sUNIX\spassword:* %n\n I get the Windows message
2009 Nov 23
2
again, nic driver order
I have two servers with identical hardware ... TYAN i3210w system boards with dual intel gigabit interfaces, and a PCI intel gigabit nic. I'm running Centos 5.4, x86_64, 2.6.18-164.6.1.el5 Every other time I reboot, the nics initialize in a different order. anaconda had setup /etc/modprobe.conf with alias lines for the cards: alias eth0 e1000 alias eth1 e1000e alias eth2 e1000e However,
2004 Dec 18
4
Free World Dialup and Asterisk
Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD
2003 May 24
4
Free World Dialup behind NAT
Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients