similar to: oddity with CISCO CCM and Asterisk

Just upgraded to the current stable ver. when I start asterisk with -vvvvvcg I get the following error [pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead Dec 23 19:25:33 WARNING[1633]: loader.c:440 load_modules: Loading module pbx_loopback.so failed! Asterisk

Hi, I was able to work out SIP trunk between Asterisk and CCM 4.x without any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk. Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not replying. For the same reason Asterisk is marking it as UNREACHABLE. Anybody got Asterisk and CCM 5.x intergation working. How can I fix the problem which I'm facing with CCM 5.x?

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

I've got problem with connecting asterisk to CCM. Our side has Asterisk system other side CCM , ehrn i dial a number on other side channles created , connections established but nothing happend , just silence , and after some time busy tone. Sides sending ad reciving g711 codec , but we need that sides send and recive g729 (we have licenses) , if in h323 conf i try to : disallow=all ,

Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, Asterisk is running strictly VoIP over the network and using CCM as the trunk. Calls from the SIP phones connected to Asterisk work fine. They can call both external numbers and any Cisco extensions attached to CCM. Calls from CCM to Asterisk fail without any notification in Asterisk (and I DID have this working at one

Hi, I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each,

Okay, another Cisco related issue (sorry!). Single Asterisk box at location 1. Single Cisco box at location 2, however the Cisco is also the PBX for location 3 (same physical machine, calls routed via VoIP). Trying to have Asterisk be able to call EITHER Call Manager location. The single SIP trunk in CCM (version 6.1 mind you) only allows a single device pool to be selected. So configuring calls

I have been getting these for awhile now in my log files. Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_spool.so' (in

Hi, I have used basic Asterisk as a PBX controlling few extensions. My question is simple, at work there is an existing Call Manager/PBX and all which manages all the extensions for SCCP VOIP phones. Can Asterisk be used to manage just 1 VOIP phone and still can make internal calls to the other extensions? Thanks, Jorge -------------- next part -------------- An

Hi, guys. I''ve been trying to use the pci-phantom command line options to xen so as to work around the hardware issue with the Marvell 88SE91xx SATA controllers in IOMMU ([Intel:] VT-d) mode, but I cannot seem to get my head around it. From having had a glance here: http://xenbits.xen.org/docs/unstable/misc/xen-command-line.html and in particular the syntax described as such:

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 I have found in http://lists.digium.com/pipermail/asterisk-users/2004-July/056111.html (Hack to make * -> (H323) -> CCM -> IOS GW work) that i need a special version of chan_h323, because of the External RTP problem. Do you know exactly which version is it? Or do i need an unofficial patch? Thanx Andr?s

Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages

Dear R-List, I would like to write nicely the names of some isotopes on a plot. The code bellow works fine. plot(1:10,1:10) text(c(2,4,8),c(2,4,8),labels=c(expression(italic(phantom(0)^{78}*Ge)), expression(italic(phantom(0)^{137}*Cs)), expression(italic(phantom(0)^{129*m}*Te))), cex=3 ) But, since I have a lot of

Dear R-List, I would like to write nicely the names of some isotopes on a plot. The code bellow works fine. plot(1:10,1:10) text(c(2,4,8),c(2,4,8),labels=c(expression(italic(phantom(0)^{78}*Ge)), expression(italic(phantom(0)^{137}*Cs)), expression(italic(phantom(0)^{129*m}*Te))), cex=3 ) But, since I have a lot of

Hi, I'm trying to make a legend with a justified list of numbers, so I thought I would use phantom(0) to align the 3-digit numbers properly with the 4-digit ones: legend(x, y, xjust=1, yjust=1, lty=c(1,2,3,4,5), adj=c(0,0.5), legend=expression(phantom(0)*300*plain(K), phantom(0)*550*plain(K), phantom(0)*830*plain(K),

I realize this is slightly off topic, but I've noticed recently that I've been unable to preform RBL lookups using the the zen and sbl spamhaus RBL lists. Currently using Zimbra Collaboration Suite using CentOS 5 I'm seeing logs showing the following output. [root at phantom ~]# cat /var/log/zimbra.log | grep spamhaus Jul 18 13:07:12 phantom postfix/smtpd[27001]: warning:

The hack below is for OpenH323, not Asterisk. This is not an Asterisk problem AFAICT. I am posting it here so that any other Asterisk user with a similar problem might benefit from it. I may or may not post it to an OpenH323 list, but since both variants of the H.323 channel in Asterisk use non-current OpenH323 versions, it may not be of any benefit to anyone anytime soon if I went that route!

There's a phantom domain master browser showing up in my Samba nmbd.log file. I keep thinking that maybe it is left over in one of the files that I transferred over from the old server to the new server and it isn't clearing itself out. Is there a way to clear that and is it possible to have a phantom browser fighting over the Domain from a copied over file? I transferred all of the

Greetings: I would like comments on this example and after fixing it up, I need help from someone who has access to insert this in R's help page for plotmath. I uploaded a drawing http://pj.freefaculty.org/R/Normal-2009.pdf that is created by the following code http://pj.freefaculty.org/R/Normal1_2009_plotmathExample.R This will be a good addition to the plotmath help page/example.