similar to: Cisco IP phone G.723

Hi all, I bought a couple phones for really cheap just for a simple solution. I'm trying to get a few 7910 to work with *. I'm just not sure how to get them to work. The 7910 just sits there "configuring IP" Here is a copy of my skinny.conf. the extensions.conf is default. I just want to bring the system up in default before a start making changes. Do I need to make

Can skinny work behind NAT? I have a Cisco 7910 using SCCP behind NAT that has one way audio. The called party cannot hear the calling party who's using the 7910. skinny.conf ; ; Skinny Configuration for Asterisk ; [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 0.0.0.0 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

At the present time you have to have a VALID ip address in bindaddr for Skinny to work. If bindaddr is either 0.0.0.0 or simply commented out all packets requiring the IP address contain 127.0.0.1. I forgot their nick, but someone in IRC recommended we make Asterisk be smart enough not to pick that interface, but I'm not sure of that is the problem or not. I simply have not had the

Is there any way to program a DSS or BLF key for another extension on a Cisco IP phone? I am using 7960 w /sip and a couple 7910's with skinny. I would like to see if another user is on their phone without the web interface. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

I am replacing a dead pbx with *. There are four lines I will be using. There is a Cisco 827-4v already in place so I will move the lines from the pbx to it. I am working with Cisco 7910 phones and I understand they use the Skinny/SCCP protocol. I am not sure if I should use chan_skinny or chan_sccp? However my main question is with communication. Do I need to use the same protocol between the

Hello all, I am trying to upgrade the firmware on my cisco 7910 without using CCM. I was told that chan skinny is possibly capable of doing that and would like to make sure. I have P00405000600 firmware which I have put in version in skinny.conf. the phone basiclaly stops at verifying load. tcpdump shows nothing happening apart from small amount of traffic to port 2000 (skinny). Does anyone

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable

I have three cisco 7910 phones connected to * through skinny protocol. When one of the phones is called, and the phone is ringing, you can hear what's going on in the room even though the caller hasn't answered. It's crazy and very hard to ignore when someone is calling :) God forbid you should cough while the phone is ringing. C.

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

Here's the problem, I upgraded all of my 7960 phones to SIP. Now my boss wants to carry his phone with him between offices. The other office has CCM which is set up for Skinny. Now I have to put SCCP back on a 7960 phone and it won't take it. Does anyone have an example of config files for sccp. Are they the same as config files for SIP? I've never had to go back to Skinny once I

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Can anyone recommend a codec pack with G723 for use under Vista? I have G723 prompts (about 70 prompts totaling 1MB) needing to be converted to G711 uLaw. I tried Audacity but it doesn't have G723 codecs. I tired some google found adware free tools and websites with no success in converting G723. It does appear the old Cool Edit (now Adobe Audition 3.0 for $349USD) can do it -jason

Hi FOlks, I am using for research purposes Kvin's codecs available at http://kvin.lv/pub/Linux/Asterisk/ G729 is working very well but g723 has a very poor audio quality. I recompiled everything with gcc4 and the distro used is Slackware 11. Anyone with some experience on that? Thanks in advance for any help. Isamar Maia +55-71-9146-8575

Hello list, i have a strange problem iam using the ulaw,alaw and g729 codecs in sip.conf i have like this [general] disallow=all disallow=g723 allow=g729 allow=alaw allow=ulaw even though i am disabling the g723 any UA could able to connect to the system and then suddenly asterisk stops working gives segmentation fault and closing the process. in logs i have this messages Feb 26 16:14:51

Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _________________________________________________________________ Use MSN Messenger to send

Hi, does anybody made G723 or G729 to work with a GrandStream Phone ? I've a Cisco here and it works fine with G723, but not with my asterisk. The bandwitdh is very important, since we will have our extensions at home. I know that I have what I pay, but the phone works with cisco. Trying to use G723 or G729 Asterisk says no codec available. Does anybody have it working with any compression

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error: