similar to: SIP Phones - Power over ethernet?

Hi, trying to build the h323 channel driver that comes with asterisk works fine, but only as long as I use openh323-1.11.7. Unfortunately, that setup seems to have a bug which misguides one of the audio streams. (So while * can "hear" me, the phone remains silent.) I suppose that bug is fixed at least in openh323 CVS. At least, I got things mostly working using the external

After some more analysis of my "dropped fragment" problem, things look like this: Cisco 7940 phone -- RTP --> chan_oh323 --> Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one

On Wed, 23 Jul 2003, Yifang Dai wrote: > I wish! My company just spend a lot $$ on the shinny CCM phone system, so I > don't think I can change that easily... But if I can get asterisk to > talk to CCM via h323, and prove it's usefulness, I might have a chance > to use * in the branches... Well, good luck, then! > By the way, do you know if we can get *'s VM to

Hi Siggi, > > 7960 and then "Call Ended" on the Display (curious about that !!!). > > That seems to be normal for the 7920. I've sniffed the registration > procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's > doing the same thing. Maybe that's some odd way of testing if the > CallManager ("CCM") really works... >

Does anyone know how to customize the order of the soft keys on a 7960 running SIP? All the documentation I could find is CallManager related. Specifically, I want to move the transfer function to the first set of buttons during a call.

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

Hi All, have some decent success on the 7920 "activation" in Asterisk. Latest status: chan_skinny does NOT work with 7920 chan_sccp does WORK with 7920 (!!) however: to remove coredumping the chan_sccp just comment out the MWI (messagewaitingindicator), then it compiles fine. Then change sccp_helper.c: return "P0060302" instead of the old value. and voila: Phone is

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

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake

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

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

Some of you may remember back in May the thread on using Asterisk as a voicemail server for a Cisco Callmanager system. My own Callmanager system is integrated into an Asterisk server for voicemail (and other things). Back in May I was using H323 for integration, but since I've upgraded to CCM 4.1 I have switched over to SIP. The integration with H323 required using Call forwarding to send

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? <http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration> I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi

hi! i had some good news regarding the cisco 7920 and the internetworking with asterisk (and possibly SIP ?). Status: chan_sccp.so not coredumping anymore :-) Phone contantly in reboot loop [see below] :-( Reboot Loop means: ------------------ Phone auth's with AP Phone gets IP from DHCP & TFTP Server Phone loads OS7920.TXT Phone loads SEP<macaddr>.CNF.XML Phone loads

I'm trying to set-up Asterisk server and I would like to buy 2 SIP phones. Has anybody tried Cisco 7960G? Or 7940? What audio compressions can I use with this phone and Asterisk? Reason why I'm asking is because Cisco supports G.711 and G.729a audio compression (probobaly some tohers but they are not listed on data sheet) and on Asterisk features i found that it supports G.729 but need

Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity server? Which functionalities Cisco Unity has than asterisk could cover? How could asterisk complement the

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

Hi All, I haven't started sip traces or debug yet, but was wondering what the deal is with the CCM and reinvite, why it doesn't work with Asterisk (using 1.2.9.1). I can make calls back and forth all day with canreinvite=no, when I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to Asterisk Server 2, I get one-way audio issues. All the RTP ports are configured