similar to: DTMF Problem w/ MeetMe

I found with some more testing that you have to setup a 5 digit number (or something longer than your phone extensions) to make the voicemail work. Now the trick is making the MWI work. Rick -----Original Message----- From: Lull, Rick Sent: Friday, July 15, 2005 3:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: VM on * for CME Install Hi folks- I've

Hi folks- I've got to the point of trying to configure voice mail on the * box for the SCCP/CME phones. The phone can call the voicemail number (8500) and I can hear Allison's voice. Attempts to punch in a voicemail box number or password don't seem to register; keypad presses don't seem to be heard by the * box. The CME configuration has the 'dtmf-relay rtp-nte' command

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've this infrastructure: |voip services| -- |*| -- |cme| -- |isdn| the voip services are logged on my *, then forwarded to number 601 on cme. The isdn calls too are forwarded to 601. On cme I've a timeout X for call-forward noan (no answer) to a specific number on * (5901) that is my x-lite software client. If 5901 is

I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a

Hi, I'm trying to get a remote Cisco Call Manager Express (CME) system behind a dynamic IP address routing both inbound and outbound calls via SIP to my local asterisk server. I've got a local CME system working fine on the LAN, where the CME is at a static IP (host=10.5.7.130 in sip.conf), but I can't figure out how to get it working with host=dynamic, even locally on a test

Hi folks- I'm working on getting a test Call Manager Express system working with Asterisk. My plan is to have * support all the voicemail boxes for the CME/SCCP phones. Right now, I can call from a SIP phone to a SCCP phone and back fine. Calls go from Phone->CME->*->Phone and the reverse. Voicemail works for my SIP phones, but does not work for the SCCP phones. I tried to follow

Hello list. No intention to start a flamewar here but I would really like opinions from those who know both the Cisco and Asterisk system. I'm working for a company with 15 offices in 11 countries, offices are relatively small (3-20 people each) and most of them have a Cisco 1760 Router installed with Call manager express (CME) and 1-3 ISDN lines (2-6 simultaneous calls). We

I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Mikel

Hello, I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front of the phone and also to hopefully resolve some issues with the phones not registering after a long period. Once we upgraded the phones now display "Error Verifying Config Info" in the Status messages and will not process the

su testuser11 cd /storage/CME/test No problem. But when I try to access the same directory in windows I get these entries in my logs.... /var/log/samba/log.smbd ------------------ [2010/01/04 16:08:25, 1] smbd/sesssetup.c:reply_spnego_kerberos(350) Failed to verify incoming ticket with error NT_STATUS_LOGON_FAILURE! log.winbindd reports no errors so it seems that the SIU/UID mapping

Hello List. I am setting up asterisk as a central dialplan, voicemail and conference solution, connected to 12 Cisco 1760 Routers running Call Manager Express IOS distributed around the world. This is all done over VPN. These routers all have PSTN access in their respective country. So far all is good, and Asterisks interopability with the Cisco CME using SIP is very good, although

Hi all, I'm porting a package that I've worked on for OS X to Cygwin/Windows. This package requires a Makefile. My question is, how can I find out (or what is), the link command? Here is the OS X Makefile: RLIB_LOC=${R_HOME} F90_FILES=\ class_data_frame.f90 \ class_old_dbest.f90 \ class_cm_data.f90 \ class_cm.f90 \ class_bgw.f90 \ class_cm_mle.f90 \ cme.f90 FORTRAN_FILES=\ dgletc.f

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

Hi, My rsync is stopped working suddenly I got following in verbose and log, mkstemp failed: No such file or directory and rsync error: received SIGUSR1 or SIGINT (code 20) at rsync.c(229) my rsync code : rsync -az -e ssh --delete $HOSTTOBACKUP:$SOURCE $DR_BACKUP_DIR/daily.0 >$tempfile 2>&1 the same code was working last week, what will be the problem, how to proceed to fix?

Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323. Cisco conf: dial-peer voice 8 voip destination-pattern 2... session target ipv4:<asterisk ip> codec g711alaw no vad h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ;tos=lowdelay ; disallow=all allow=alaw allow=ulaw allow=gsm context=from-internal extension.conf [from-internal] exten =>

Hello All, We have been doing Asterisk and CME implementations recently but we almost always exlusively bring in analog lines and or PRI for PSTN access to our systems. I have known about providers providing SIP based lines and SIP trunks to end users for PSTN access. I am curious about the following: - How practical is this? The idea of terminating pstn calls to across the Internet

Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and