similar to: g729 codec over SIP Trunk between CCM and Asterisk

Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or nothing. Specifically, this is what it said: ============================================== *Note: There is

Hello, we have distributed lots of cisco spa303 IP phones and get them work with Asterisk. I have configured proxy and alternate proxy and enabled "dual registration" features in provisioning files(xml files). All phones are able to subscribe to both of servers. But the problem is, if main server goes down, i am obliged to wait nearly 20 second in order to place a call over second

I'm following http://supportforums.blackberry.com/rim/board/message?board.id=java_dev&message.id=25929&query.id=182675#M25929 to install the Blackberry Eclipse plugin. I'm stuck at step 4, as I'm not sure what it is asking. I have a basic understanding of Paths within Windows, but don't know how to edit/add/create them within Linux/Wine. Anybody know what I'm supposed

Hello, If I'm not mistaken, it is not possible to get T.38 on a SPA3102 FXO port (it is possible with the FXS port). Do you know, by experience preferably, if this is possible with an SPA8800 FXO port ? Regards

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?

I have a new dell t105(opty 1210's with 4 gigs of ram..sata hdd's and sata dvd-rom). I can bot to the Cent 5 64 bit dvd fine but once it asks where the install image is i tell it cd-rom..then it rejects the dvd spits it out and tells me it's not there. the same happens on 32 bit. I have downloaded centos5 twice now, md5 verified..and even burned at 1x(that took forever). Any

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

Hi, I''m runing a Fedora Core 6 on a Power Edge with Xeon Cranford that I planning to running Windows as a guest S.O. and reading "Fedora Xen Quickstar FC6" as document to install Xen on this server. This document recomend a "grep vmx /proc/cpuinfo" test to verify if computer is Intel VT supported. Running this command I don''t find any information about vmx.

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]

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 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 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

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 ,

Hi List, I have a system that connects into Asterisk 1.4.41 using CISCO CCM 7. Everything works great except when a call is transferred to the operator. The call goes to the operator via a native bridge and is completed, then a "phantom process" starts and tries to launch a new call every 15 minutes. I modified the dialplan to hangup these phantom calls, but no still

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

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!

OK, thanks again for all your help. I have resolved this, finally. The problem was that I configured VLAN 48 as the native VLAN on the trunk port.That was a mistake as apparently the native VLAN is the one where Cisco does not bother to tag packets. For now I set the native VLAN to VLAN 1 and that works. Cheers, Boris. On Sun, Jan 25, 2015 at 7:13 PM, Boris Epstein <borepstein at

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid