similar to: Cisco CME Integration - IOS Version known to work?

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've followed the wiki and can't get the message waiting indicator to light on the phones connected to the CME. The * console gets messages that registration failed - Jul 20 13:27:33 NOTICE[22017]: chan_sip.c:7708 handle_request: Registration from '<sip:6502$@172.28.1.34>' failed for '172.28.0.100' I get one message for each number that is setup on the CME box.

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

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 =>

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 searching for a way to inform my Cisco CME that a number on Asterisk server is busy. I have a SIP trunk between Cisco and Asterisk and some Cisco ip phone have a speed dial with a number registered on Asterisk. How can I exchange busy information between two PBX? Thanks Enrico.

A little off-topic: I have the following hardware: 2610 XM NM-2V VIC-2BRI NT/TE IOS loaded: flash:c2600-ipvoice-mz.123-5d.bin" I get the following error while booting: %C542-1-UNKNOWN_VIC: VNM(1), vic daughter card has an unknown id of FF Is the VIC-2BRI compatible with the 2610XM? What IOS needs to be loaded? http://www.cisco.com/en/US/products/hw/modules/ps2641/products_tech_note0918

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 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

First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works

-----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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I'm working on sip trunk between cme 4.x and asterisk (trixbox 1.2.3). Well, the trunk is partially working, asterisk' extensions talk with cme, but - - when cme try to connect to asterisk' number, receives "the number dialed is not in service". - - calls from ISP through asterisk to cme don't work completely,

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 All, We have a Cisco CME linked to our Asterisk PBX (named 'epstein'). Off said PBX we have numerous other PBX's, some IAX and some SIP. On a call placed from CME (SIP) to 'epstein' it all works fine except for a few quirks. When calling through epstein to an IAX peer we get '100 trying' followed by '180 ringing' sent back down the SIP leg to CME.

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I would create a structure like this: external sip server \ external sip server |-----| Asterisk |------| Cisco CME |-------| ip phones | external sip server / I would use Asterisk as SIP client for some SIP accounts on external servers ... then register those via H323 (if possible; skynny?) on Cisco CME ... Then I would use Asterisk

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

https://bugzilla.mindrot.org/show_bug.cgi?id=3196 Bug ID: 3196 Summary: [Information Disclosure] OpenSSH_7.4p1 Raspbian-10+deb9u7 discloses OS version Product: Portable OpenSSH Version: 7.4p1 Hardware: Other OS: Other Status: NEW Severity: security Priority: P5