similar to: QoS anyone?

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

Does anyone know whether the linux machine running * have to have a sound card on it in order for musiconhold to work for sip phones? I've tried about everything (including tons of google searching) to get it to work, and nothing. When a call is placed on hold between two C7960's, the CLI indicates: -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack

Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200,

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 -

Trying to test the music on hold function and can't seem to get it to work. If anyone has it running, could you give me a clue? (I have googled and found lots of questions, but no real suggestions.) I downloaded and installed the mpg123 package. From the RH9 console I can start the executable and hear the music via the speakers. The executable is located in /usr/bin. (That works!) I set the

Using the following defnitions with a C7960: exten => 3001,1,Dial(SIP/3001,15,r) exten => 3001,2,Voicemail2(u3001) exten => 3001,102,Voicemail2(b3001) exten => 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second timeout and is dropped into the 2-priority as "unavailable" (not the 102 busy as

I upgraded my 7960 to sip v 6.3 and my display time has now disappeared from the top left corner. Loadid: SW: P0S3-06-3-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38 Here is the section dealing with time in my SIPDefault.cnf file. Does anybody see anything wrong with it or have any other ideas? # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)

I'm about to download Festival source, apply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers.com

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

Hi all, I have looked at cisco docs and it says DND is set by pressing the services button and choosing DND. Does anyone know how to configure DND in the services.xml file. I've googled around and not found anything. When you enable it in SIPDefault.cnf it just allows you to use it once. Many thanks for all your amazing work. Daniel Niasoff

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.......

Hi, I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ? From http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2, I got the following: 1. Copy the desired binary image from Cisco.com to the root directory of the TFTP server. 2. Specify the image in the configuration file image parameter for the protocol

> -----Original Message----- > From: Babak Pasdar [mailto:bpasdar@IGXGlobal.com] > > I was wondering if anyone else has had this problem. I have purchused > several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run > accross 2 that that give me malformed TFTP and refuse to upgrade to the > latest version of SIP code -- 5.3. In fact some of the other phones also

Greetings to all. I am hoping someone can help me out with a problem I am having getting my Cisco phones, 7960s and 7940s, to download the appropriate files from our TFTP server. The TFTP server is running on Fedora Core 4. The TFTP server appears to be setup properly: service tftp { socket_type = dgram protocol = udp wait =

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only

Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The

I have a done google seaches on convertion and so far they all failed. Rich adamson and wheely-bin.co.uk Here is what I have Laptop running solarwinds tftp with the following files OS79XX.txt <- POS30201 SIP<phonemac>.cnf.xml RINGLIST.DAT <-Ringer1.pcm Ringer1.pcm The phone (7940) is hardcoded to the TFTP ip of the laptop and has a hardcoded ip is was on firmware version:

Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 mike@introspect.com www.introspect.com

Hi, Anyone know what the settings in SIPDefault.cnf should be for Cisco 7940 phones this year? Came in today to find they'd all moved one hour ahead (NTP server is correct and ok). Found the "day" was set to "26", but on trying to change the settings to the below, my test phone isn't changing back: dst_start_month: March ; Month in which DST starts dst_start_day: