similar to: sip trunk - SIP/2.0 488 Not Acceptable Media

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 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 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

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

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

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

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

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, 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

-----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

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

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.

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

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've installed dovecot 0.99.14 (last port on my freebsd system), and now I've to migrate from bincimap. My system uses qmail with vpopmail (uid/gid 89) and Maildir format. Well, now the authentication is ok, but when I try to connect to mailbox I receive an error like "IMAP command SELECT failed". My dovecot.conf:

I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering

Folks, I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via standard wic-t1 card. The NEC needs to call two different asterisk servers with 4 digits. I have two way calling working with the one * box, but the other is perplexing me. Here's the layout * <--> Cisco 2811(192.168.13.1) <--> 1.54 point to point <- Cisco 3725(192.168.8.1)<-> NEC 2400. The