similar to: Cisco 7960 (SIP) with Asterisk: how to get # to work during a call

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -

ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck.

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

I have Asterisk version 1.0-RC1 running on Debian Woody. I have 1 analog phone working, 2 inbound lines working, X-Lite is working. The problem that I am having is with Cisco 7960 with SIP version 7.2 software. I can make outbound calls and they work fine, I even get a notice that I have voice mail on the phone and it seems to register properly but I can seem to dial to the phone.

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have

Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47

I'm having an issue sending DTMF to cisco dialing this extension I should hear the dtmf tone RTP playload 101 has been sent to the cisco phone, but no audio. in the dialplan exten => 8603,1,Answer(1) exten => 8603,n,sipdtmfmode(rfc2833) exten => 8603,n,SendDTMF(1|100) exten => 8603,n,hangup() sip.conf dtmfmode=rfc2833 SIPDefault.conf I did play with all possible settings for

I have noticed a problem with the Cisco 7940/7960 phones where if you put your voice-mail box on hold using soft keys and come back you can no longer navigate. I am curious if anyone else can duplicate this problem. Happens reliably for me with the 7940 phones. I use rfc2833 for DTMF. I would think it was a Cisco bug, but for the fact that this did not happen with older version of

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

Hi Jean-Marc, thank you for that suggestion! It seems that the file "fixed_c6x.h" is not part of the Opus sources, so the compiler cannot find it after enabling the TI_C6X_ASM config option. Maybe it was only part of an early version of the Opus sources? I looked for the file in versions V1.1, V1.1.1, V1.2alpha and V1.3 but did not found it. Do you have an idea, where I can get the

Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike

Dear Opus family, we have implemented the Opus codec at a Texas Instruments DSP C6418. It is working fine! Does anyone has experience with the configuration of the codec for a speed optimized implementation on that DSP? At the moment, we use the following settings: #define NONTHREADSAFE_PSEUDOSTACK 1 #define FIXED_POINT

All, I have upgraded by home machine from Fedora Core 5 (FC5) to the recent FC6 and am struggling to build Zaptel-1.2.10 and Asterisk-1.2.13 on the box... which is an Intep P4 2.8GHz HT processor box with 845 chipset, hence the kernel installed is 2.6.18-1.2798.fc6-i686 so we hve this: [root@gate zaptel-1.2.10]# uname -a Linux gate.tubby.org 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:54:20

Hi Jean-Marc, Alfred and Greg, Are you receiving the mails from IETF about draft-ietf-avt-rtp-speex The mails are not coming from AVT mailing list, but I think we are all 3 part of a minimal list (draft-ietf-avt-rtp-speex at tools.ietf.org) dedicated to latest discussion about the draft. I have answered some questions, but there are small changes and adaptation still required to the ietf

Hi all, Below is the Vorbis RTP Internet Draft as sent to the AVT working group of the IETF. Comments and feedback is still welcomed from the Vorbis community. Cheers Phil ---------------------------8<-----------------8<------------------------ Network Working Group Phil Kerr Internet-Draft The Ogg Vorbis January 07, 2003 Community / OpenDrama

Hi all, Apologies in advance, this email is quite long. I've prepared an updated Vorbis RTP Internet Draft, which is a continuation of draft-moffitt-vorbis-rtp-00.txt which can be found below. If this new draft gets the ok I'd like to submit this to the AVT WG later this week. There are a number of changes over the original I-D, notably the changing of the M bit function in the RTP

Peter Saint-Andre wrote: > On Mon, Jun 04, 2007 at 03:58:17PM -0700, Ralph Giles wrote: >> On Mon, Jun 04, 2007 at 03:20:05PM -0600, Peter Saint-Andre wrote: >> >>>>> Hi Luca, >>>>> >>>>> Thank you for working on the Theora and Vorbis drafts in the IETF's AVT >>>>> WG. Do you also plan to work on the Speex draft?

On Mon, Jun 04, 2007 at 03:20:05PM -0600, Peter Saint-Andre wrote: > >>Hi Luca, > >> > >>Thank you for working on the Theora and Vorbis drafts in the IETF's AVT > >>WG. Do you also plan to work on the Speex draft? > > > >other people are working on the speex I-D, Ralph should be able to tell > >you which to contact ^^; > > Hi Ralph,

I am transferring some files from a RH7.0 machine to a RH7.2 machine as root with a cron job, using the following: rsync -avt /home homeserver::home rsync -avt /etc homeserver::etc Some of the files transfer OK, but I get: building file list ... done failed to set permissions on home : Operation not permitted home/brad/ home/ftp/bin/ home/ftp/etc/ home/ftp/pub/ home/rsyncuser/ failed to set

I'm having problems with a new asterisk PBX install. the phones/ATAs are all linksys/cisco. They all worked before with a commercial softswitch. Most of the linksys devices offer auto, inband, INFO and AVT. I'm looking for suggestions. Thanks in advance -- One day at a time, one second if that's what it takes