similar to: CIC Code

OK here it goes.. Caller ID is two parts or actually three: Part 1 Number only Part 2 Number + Name Part 3 Whole lotta stuff (also known as ADSI) Here is the US, I cannot speak for other countries. When party A places a call to Party B. Party A's Telco picks up the number, either from a table on the switch or passed from the PRI from Party A. Then on the far side (Party B's Telco)

I am using an Asterisk box as a mini-softswitch and have run into a minor (hopefully) road block. The far end switch requires CIC (Carrier Identification Code) in the SIP invite like: INVITE sip:+18001234567;cic=+16789@hostname.com;user=phone SIP/2.0 ^^^^^^^^^^^^^^^^^^^^^^^ Is there a way to configure Asterisk to send this in the SIP invite? Any help would be

mdew:~# tc qdisc add dev imq0 handle 1: root htb default 12 r2q 1 Cannot find device "imq0" mdew:~# lsmod Module Size Used by Not tainted ipt_REDIRECT 728 0 (autoclean) ipt_MARK 728 2 (autoclean) iptable_mangle 2100 1 (autoclean) ipt_REJECT 2712 4 (autoclean) iptable_filter 1672 1 (autoclean)

Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. >From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no

zaprtc 'load' is failing on my machine (the make was fine, same output as other posts to this list) [xxx@xxx zaptelrtc]# make load sync modprobe zaptel insmod ./zaprtc.o ./zaprtc.o: init_module: Input/output error Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or

Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674

Since updating * via CVS earlier this week, I've been having problems with cdr_mysql. Prior to that time my queries and cdr all worked fine. Now, even though my queries still work, I get the messages similar to this: ERROR[1211374384]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas -- Hungup 'Zap/74-1' Anyone have any suggestions what went wrong?

Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Notice which appear when voice prompt is played. NOTICE[1217602880]: File sched.c, Line 209

I have had a web server listining sql-1433, www 80, ftp-21 using proxy arp with sub-netting in a three interface DMZ. All these ports are in the rules file as ACCEPT. With one exeception that 1433 allows a few host from the net. 21 and 80 allow all net to dmz connections. The policy is DMZ to net ACCEPT This has been working great for about a month or more until I rebooted the

Hello everybody, After going through multiple posts on Internet and trying different things I can't seems get zaptel wcfxo loaded on the server correctly. I run dual P3 500 Mhz box. After reading README.udev I put a file 60-zaptel.rules /etc/udev/rules.d # Section for zaptel device KERNEL="zapctl", NAME="zap/ctl" KERNEL="zaptimer",

I have 1 * box having x100p installed and the other has no zaptel card at all. both of the * box has compiled in ztdummy module and both have been activated by modprobe ztdummy. When using trunk to connect the 2 *box. The one without zaptel card complaint about unable to accept trunked packet: no matching peer. On the one has zaptel card I have tried to remove the ztdummy module and connect

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here

Hi, Is there any (easy) way to get Asterisk to include CIC-information in the SIP INVITE? CIC: http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/chapter3.htm#1314580 I need my SIP INVITE to look something like: INVITE sip:5550001;cic=+16789@172.18.202.60:5060 SIP/2.0 I'v tried a couple of different things but can't find anything

I wonder if there is an application available, what would allow me to have a conference call (meetme) with video.

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

I've got two patterns I want to match on making an outgoing call... (one day - to do Least Cost Routing for Cell/Mobile calls) Firstly - I prefer '0' rather than '9' to get an outside line... Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084) or its just another number to dial... I added the following... the playback just advises me which 'route' is

I have set up a "community type server" for Caller Name storage and lookup. IT IS NOT SS7 BASED....(not yet) Currently it supports five switches. This is a proof of concept. If you are receiving calling number, but not name, and would like to be involved in this project, please e-mail me off-list. tchandle at bayou.com I plan on running this for about six months, then make a

Hello, I'm trying to run smbtorture against another system. I have installed version 4.0.0alpha9 locally. The remote system is registered with ADS as: distinguishedName: CN=bl-uits-cictest,CN=Computers,DC=ads,DC=iu,DC=edu name: bl-uits-cictest dNSHostName: bl-uits-cictest.ads.iu.edu servicePrincipalName: HOST/bl-uits-cictest.ads.iu.edu servicePrincipalName: HOST/BL-UITS-CICTEST The

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have