similar to: Cisco 7912 Phones & XML

Adam Mattina Networking & Systems Support Layer 8 Group, Inc. 585.442.8888 amattina@layer8group.com <mailto:amattina@layer8group.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061109/023a0a3e/attachment.htm

Salve, pensavo che potrei passare a farle vedere le nuove funzionalit? che ho implementato luned? mattina, lei sarebbe disponibile? -- Maurizio Marek idnondisponibile@fastwebnet.it

The documentation of how to use the 79xx series' phones and features with Asterisk is really hard to find and put together. The higher end phones like 7970 are more like converged PC+phone, a thin client to telephony and network apps. But it's really hard to target it as a development and deployment platform because the docs and techniques are so obscure. There seems to be a fair amount

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry

Hi ! We have Cisco 7912 phones, and the doc says that I can "create up to four speed dial buttons on my phone using the Cisco CallManager". Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D

Hello all, I tried to set up some cisco 7912G phones with asterisk with not much success. Can anyone give me some tips on this one ? Is it better to use SCCP or SIP ? Where can I get a SIP software image ? Regards Kiss Karoly

Hello all, I feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for "7960 sip firmware" and find that fast). Thanks for the help, Jeb Campbell jebc@c4solutions.net

Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert

Are the 7.x SIP images for the 7940/7960 compatible with the 7912 or no? I ask because there is a 'special' section for 7912 SIP images that seem to be on a totally different revision train. -- GPG/PGP --> 0xE736BD7E 5144 6A2D 977A 6651 DFBE 1462 E351 88B9 E736 BD7E

Anyone know what the file format for the cisco 7912 RINGLIST.DAT file is? I have a RINGLIST.DAT file for a 7960 that is working fine. It's just a list of the ring name and the file name but it doesn't work for the 7912. I've read somewhere that the 7912 needs the ringlist.dat file to be in xml but I can't figure out how exactly it is constructed. Thanks. -- -------------- next

Hi, I am using a cisco 7912. I setup the phone at my first location. I edited the gk<MAC>.txt file setup the proxy and UID etc... values. generated the gk<MAC> file and booted the phone and it worked... I then mailed it to its final destination. this place has other 7912 phones working there.. I copied the gk<MAC>.txt file to that server, changed the proxy and UID values,

I have a 6 - 7912's. I have a TDM2402E echo cancel card. Asterisk 1.2.12.1 Some times extension to extension the audio is only heard one way. Some times extension to TDM2402 calls audio is only heard one way. I have turned off the echo suppression on the 7912's config. Any idea why it would be doing this? Something to try? THanks, Jerry

Not a bug in a very strict sense, but still something to be fixed eventually: > s <- sapply(0:7, function(i) rawShift(charToRaw("my text"),i)) > s ... nothing is printed at all > str(s) raw [1:7, 1:8] 6d 79 20 74 ... > c(s) [1] 6d 79 20 74 65 78 74 da f2 40 e8 ca f0 e8 b4 e4 80 d0 94 e0 d0 68 c8 00 a0 [26] 28 c0 a0 d0 90 00 40 50 80 40 a0 20 00 80 a0 00 80 40 40 00

Hi all, I playing around with some C7912 IP phones (SIP FW). They work nice with asterisk, but I found the following issue: o When I configure the voicemail number (8500) to access VM I can push the "messages" button on the phone to access my VM o The phone can setup DND and call redirection on NA and BUSY o MWI is also working fine ;-) But in this case the call will be

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List