similar to: CAPI<->SIP broken incoming audio

G'day all, I've been googling myself silly looking for help on this one but have come up blank. I have an AVM Fritz!Card PCI, and I'm using chan_capi v 0.3.1 with * from CVS-HEAD-05/08/04-22:48:00. I can start * and make and receive calls on ISDN fine but after a few hours of * uptime, on any ISDN call I make or receive from my SIP handsets (7960 or ATA-186) I get bad audio: on the

Those of us that use Cisco 7940/7960 sip phones know that we've been impacted by two very different changes that have occurred over the last couple of months. First, when cisco created sip v6.x code, they implemented a new DSP (as well as other software changes) that effectively drops any incoming rtp packet that does not have even timestamps within the rtp packets. The dropped packets

I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? ---------- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part --------------

To get the SIP firmware for these phones, you need to buy a Cisco SmartNet support contract (about $75 USD in the USA, though I've heard rumors a Europe-only contract exists for about $10 USD.) You can purchase one through most Cisco resellers. That will give you access to Cisco's download site. Configuration of these phones under SIP is not quite as straight forward as it is with SCCP,

Hi everyone, I have the following configuration: Calls are coming in into the Asterisk system via an ISDN card (AVM ISDN-Controller C4) When a call is received, it's placed into a hold using the dial-plan: exten => s,1,Wait,2 exten => s,2,Answer exten => s,3,SetMusicOnHold,random exten => s,4,MusicOnHold,random An application is running in the background, using the

Good day, The unusual thing about this problem is that it doesn't occur just during a CAPI call, or just during an IAX/SIP call. Only during IAX/CAPI I'm having some trouble with the CAPI interface and it only occurs when a call comes in on an IAX channel and goes out the CAPI interface. The capi debug in the asterisk console is below as well as the relevent parts of .conf files from

I concur with Conrad. Cisco phones were retrofitted for SIP, whereas Snom phone are built around, and expressly for, the SIP standard. To be in compliance with Cisco regs, you are also supposed to have a SIP User license and a Smartnet contract for each phone if you abide by their program. I realize it is not difficult to obtain the SIP firmware, but if you were dealing with a Cisco authorized

Hello, i have asterisk with chan_capi ( AVM Fritz card ) working, isdn line BRI, setup to point-to-multipoint. working fine, but when both channel on interfaces are used and somebody wants to call in - the response is "ha-la-li" - sounds like msn or called number does not exists. i think that right response should be - line busy tone... right ? can anybody tell me what is wrong ?

Hi, We've just bought a new cisco 7965g and web are trying to connect it to asterisk. I've bought smartnet and downloaded cmterm-7945_7965-sccp.8-3-3SR2.exe cmterm-7945_7965-sip.8-3-3SR2.zip The zip file contains: Archive: cmterm-7945_7965-sip.8-3-3SR2.zip Length Date Time Name -------- ---- ---- ---- 2496963 11-05-07 10:06 apps45.8-3-3ES2.sbn 585536

Hello all, I've just finished to install chan_capi with 3 AVM Fritz PCI cards. It correctly loads the 3 drivers, and * starts without errors. immediately after * start, audio quality is really fine, but, after the first incoming call, all incoming audio is broken, trembling and stuttering.

I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Mikel

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, 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,

Just a quick note to people looking for SIP firmware images for Cisco phones: To access these files from Cisco's website, you need to have a Service Contract (SmartNet) on at least on of your phones. I though a contract was several hundred dollars, but it is way cheaper! Their lowest level, 1 year Next Business Day 8x5 SmartNet program, costs around $70.00, and gives you unlimited access to

I have found some references to the "hint" (or HINT?) variable and method in the source code, but quite a bit of Google-ing has not turned up any extensive answers as to some real-life examples of how to use this perhaps very useful tool. I understand the point of the tool, but I need to get some actual configs to look at before I think I'll figure it out. Even if my

Hello, I have installed an asterisk server with 7960 sip phones and an eicon card (diva 4bri server, I use only three channel) using capi channel. Everything seems to work fine, but from time to time (two or twice a day) during a call I get a capi error in asterisk log and asterisk doesn't work anymore: ==== Apr 1 10:38:24 ERROR[20495]: File chan_capi.c, Line 738 (capi_write): error

Hi everyone, i have a problem on a running xen installation. The server on which the following problem occurs manages 7 VMs at the moment. Now I wanted to create a new one. After sending the xm create command I get the following error: Using config file "/etc/xen/de-smartnet-asterisk". Error: Creating domain failed: name=de-smartnet-asterisk In my xend.log I can find the following

a) jerjers been doing a lot commendable work for * b) support is not mandatory, and i agree with royk it should not be withheld based on political viewpoints, that's pointlessly draconian c) choice is always good, so people should have the option of oh323 or h323, let them decide, and not limit them, unless astmaster chooses to limit them, and that too based on valid points d) jerjer gave a

Thanks everyone for your feedback, especially Mark. I now have the ALL the files I need. My order still stands for the $8.00 product from CISCO but the CP7960 dealer sent me all the files. Now I will move on to completeing the setup of the TFTP server. Thanks again -----Original Message----- From: Michael Loftis [mailto:mloftis@wgops.com] Sent: Friday, February 18, 2005 7:51 PM To:

Hi, I?ve strange problem when I?m making a call from SIP (Cisco 7960) to capi (Fritz PCI). When I call a national number, I?m hearing the ringtone when the called party is ringing but when I call an international number, I don?t hear the ringtone and I?ve a silence until the called party answers. Both call are going through the same extension. Here?s 2 log files, one with a national number and