similar to: SIP 7960 soft key customization?

Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after

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... >

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

Are there any cheap SIP phones (like the Grandstream for example) that support power over ethernet? What is necessary to support SIP phones in a Cisco Call Manager environment?

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

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

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

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 there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE

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 Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

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

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 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

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

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

* compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled --

I assume there are several people on this list that have Cisco Call Manager implementations under their belt.... We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two

Anyone have the 7970 xml config for sip yet? Aaron