similar to: Codecs Problem?

Check to see if you can get a IOS code leverl that supports SIP on the 6500. then maybe you can use your E1 card directly. you can also get a SIP version of the code for the 7960's etc Dave >>> jwi@weball.csis.dk 6/28/2003 2:56:12 PM >>> Hi Chris I've done a lot of things with Cisco AVVID solutions in the past. > CallManager).....am I right in saying that Cisco

I have a Cisco 2600 which understands the "gsmfr" codec, which appears to be what Asterisk calls "gsm" -- at least it ends up using it. I also have a PSTN gateway which is speaking ulaw. When the 2600 calls through Asterisk to the PSTN, it negotiates the g711ulaw codec, but when the PSTN calls through Asterisk to the 2600, it seems that Asterisk is doing translation, and it

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com

Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

Hello, I am trying a setup that is the following: SIP Phone (Zultys) --> Asterisk ---> H.323 GK (Cisco) ----> PSTN Any calls from H.323 GW through GK goes to PSTN, no problem. SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem. SIP Phone to PSTN, rings normally, on the PSTN, then connects when the PSTN phone picks up, no audio on both directions. PSTN GW support

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)

On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote: > Does anyone have a way to do wake calls? > > > > Jordan Novak > > Communications Technician > > Logistics Health Inc. You could use cron and /var/spool/asterisk/outgoing scripts to dial numbers, etc... > Can you elaborate, I am fairly new to Linux and a phone guy to boot. I am looking for a way for the

Topology: PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server When I make a call to a VoIP user from the PSTN, the call gets routed through the PBX, and Cisco. Because of that the DTMF tones are passed inband, which I can hear on the VoIP end of the call. However, I have one extension on asterisk set up so that I can check voice mail when away from my

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

Hi All, I'm trying to get a Cisco 7905g IP Phone to work with our Asterisk server but I'm having problems getting the phone to answer a call or make a call. I'm using the stable branch of the asterisk CVS on a RH9 box. I have got the phone to register with * and it retrieves it's extension number, date & time etc., but when I pick the handset up it's just dead silence

We're using Cisco phones running skinny protocol. When I call other extensions I don't get a ringtone, although the remote end does ring and when answered we get clear two way audio. When I call a queue from a skinny phone then I don't hear the announcements. Likewise we don't hear music on hold on these phones, although we can see mpg123 in the process list and ls -l the fd

Hi guys, This is my first post in this list. I'm newbie with Asterisk and VoIP technology. First, this is my scenario: - Asterisk server (asterisk-1.0.0) - Cisco router connected to PSTN I've got redirected incoming call from extension 532x from Cisco to Asterisk server. Asterisk server H.323 channel receive the routed call from Cisco, and it launch the following dial plan:

Hey, Does anyone know how to use the cisco router with an fxo wic with Asterisk? I don't have enough space on this device to support an IOS that supports sip or h323. Currently the only one signaling in there says Cisco. I assume this is the skinny protocol. Does anyone know how to configure this 2600 with Asterisk? Thanks, Erik

I just got a Cisco 7960. Of course, it has the Call Manager image on it, and I don't have access to a SIP image. So, I read that Asterisk has 'some' support for the skinny protocol. I played around with it, and I actually got the Cisco phone to display line 1 on it. I can call another phone (SIP) from the cisco, but I cannot hear on the other phone. The cisco, however, can hear the

First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like to use with asterisk, I have set them up using chan_skinny. The phones work well, except the only problem is that it is like the cisco phones are muted. When I talk on the cisco phones I can hear my self through the ear peice, but the person who I am calling can not hear me at all. I have tried various cisco phones from various

I have Asterisk up and running and it is working great with my SIP phones. However, I have some "Skinny"-protocol Cisco 7960s. Does Asterisk support the Skinny protocol? I've seen some references to Skinny in the software. If so, should I stick with Skinny with the 7960 or convert to SIP? If anyone has some Skinny confs they would send me I'd be much obliged. If I should