similar to: SIP with CIC

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

Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the

USING: rsync version 3.0.6 protocol version 30 on a Sun Solaris 10 x86. This is a precompiled version from opencsw. /opt/csw/bin/rsync -n -axzH -v --delete-after --log-file=$RSYNCOUTPUT/export_-${DAILYDT}X --rsync-path=/opt/csw/bin/rsync -e "ssh -i /root/.ssh/id_hertz" --max-delete=100 /export/.zfs/snapshot/$DAILYDT/ hertz3:/zvol/backup/gauss/export When I added the

It was pointed out that this information is in the --itemize-changes parameter in the man page. I looked at the man page but missed this part. I guess it was the forest for the trees. Thanks and it is resolved. Robert USING: rsync version 3.0.6 protocol version 30 on a Sun Solaris 10 x86. This is a precompiled version from opencsw. /opt/csw/bin/rsync -n -axzH -v --delete-after

Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. Thanks in Advance, Jason Miller

Hi, i'm trying to setup Asterisk to work with GnuGK using the Openh323 channel driver. I have a Gatekeeper that gets H.323 calls from a Cisco GW. To this Gatekeeper I've registered some endpoints, Cisco ATA186, Snom 100, etc. Now i want send the numbers 083xxx into Asterisk. Easy, i'll just enter something like this into oh323.conf: gwprefix=083 And all my calls starting with 083

Overlap receiving timeout, plus dialplan latency, causes network to retry SETUP https://issues.asterisk.org/view.php?id=16789 This patch removes the requirement that some may have found that you need to insert a Proceeding() statement very early in your dialplan, otherwise an inbound overlap call may retry and fail. Our experience was from a PRI connected PABX, if we took too long doing

Hi All, I need to parse data from a file, example shown below. The first two lines can be skipped, the third line contains the column names. The next 13 lines can be skipped. The next line "1991" is a year value, with the following 13 values data for that year. The file then repeats this format with (year, 13 lines of data for that year). I would ideally like to end up with an

All, I have been beating my head against a wall trying to figure out how I would implement a separate moderator code and participant code for the same conference using meetme, the deal is I dont want the participants to be able to join until the moderator is in the conference. Is it possible to do this using the apps as they are , or is their a way to use an Agi script, is that the only way?

Hello, I saw that with the library scriptaculous it is possible to create a system of tab. Only I am a beginner. Somebody can to help me to use this library in rails? Thank you in advance.

Thanks, Mark! Here's a summary of what one needs to do in order to get NAT working with Asterisk. Please note that I have a Cisco ATA-186, and your experience may be slightly different based on the equipment you're using. You'll need to have a CVS updated version of Asterisk as 2003-03-06 ~2:00 PM EST. NOTE: This currently works for outbound calling only, not inbound. In other

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP

http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/atarn/atarn3_0.htm LOVE IT.. call transfers work better but not totally as expected yet. I'm still tweaking the configs. Also bitaid will help alot. bkw

Hi, An Cisco 7960 ipphone has been set to SCCP firmware by one of our students. I want to set it to 7.5 SIP firmware and I've been unsuccessful yet. Firmware versions are SCCP 3.0 (Source: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1045789) ie: Application Load P003F300 Boot Load ID PC030300 When I browse, phone settings, I see

hi guys, found this logs on my mail server about possible fraud attempt and phising. is this normal ? Found ip-based phishing fraud from 10.2.0.0 Found ip-based phishing fraud from 255.255.255.255 Found ip-based phishing fraud from 10.1.0.0 Found ip-based phishing fraud from 255.255.255.255 . MailScanner has detected a possible fraud attempt from "ee.ee.ee.ee" claiming to be

created you must place it in your web directory on the server. =20 I chained the command and also wrote the output to an xml file in the web directory. The command looks like this: =20 'php /etc/asterisk/directory.php.txt > /var/www/html/directory.xml' =20 System Speeddials using Services Button =20 =20 For speed dials I modified the php code to look to a specific file in the

I have just got a Cisco 7941G and am experiencing the exact same problem (phone is requesting .tlv file from TFTP server and never asks for .cnf.xml file). The phone originally had SCCP on it, but I downloaded and flashed with the latest Cisco SIP image (8.4(3) released 2009-01-13). In reading your message below, it looks like you were going to try an incremental upgrade?did you have any

file is created you must place it in your web directory on the server.<br> &nbsp;<br> I chained the command and also wrote the output to an xml file in the = web directory.&nbsp; The command looks like this:<br> &nbsp;<br> &#8216;php /etc/asterisk/directory.php.txt &gt; /var/www/html/directory.xml&#8217;<br> &nbsp;<br>

Hello! I would first like to appologice if this question does not fit on this mailing-list. I am new to Linux and I tried to compile R on my Ubuntu Warty linux. I followed the instructions in "R Installation and Administration" manual. It seams that there was a problem with "configure", since when running "make" afterward gave me this reply: make: *** No

Hello, I just can't seem to get the MusicOnHold function to work out ok. I' managed to get the MP3Player app to work out fine, but when I run the MusicOnHold all i get is siliece. I can see that Asterisk executes mpg123 properly (I think) #ps axuww|grep mp gk 4383 0.0 0.4 3736 552 pts/4 S 15:06 0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 sample-hold.mp3