search for: shawnl

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Has anyone had any luck getting an asterisk box to talk to a Nortel C15K softswitch? Or any Nortel "sip" products? I've been playing with it for several days and can't seem to pass calls either direction. I know that whike the Nortel says the C15K speaks SIP, it really speaks nortel's implementation of SIP, but I thought I could get it to at least pass simple calls back

I'm wondering about the difference between Cisco Call Manager and SCCP(2) network traffic. I'm working on getting a Cisco 7960 phone to speak through a NAT to an asterisk box, without having to do a bunch of port forwarding on the NAT device. Without the nat, everything works fine. If the phone is behind a cisco pix that is doing the natting, it works fine (fixup protocol). If the

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

This one is a little off-topic, it's more about the phone than asterisk itself. I have a cisco 7960 configured with 2 lines to 2 different sip providers (cant get asterisk to register with the 2nd provider, but that's another story). Is there a way yo determine which direction speed-dial buttons will go out? I'd like to have speed-dial buttons that will go out on line2 instead of

I'm in the process of writing a simple autodialer to dial a list of numbers and play a message. One of the options I want to give them is a way to "dial X to have a customer service representative call you" Looking for a simple way to pass the number that I dialed to a script in extensions.conf... something like this: [serviceinterruption] exten =>

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just

Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no

...ot; Content-Transfer-Encoding: 7bit What LAN and you using? ELAN or HSP Are you trying to connect to a signaling server? Please provide Nortel config. Jonn -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of shawnl at up.net Sent: Wednesday, November 28, 2007 2:06 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Asterisk <-> Nortel Phone Switch Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so...

Does anyone have a small, plain services.xml file for a cisco ip phone, preferably one that will work on a 7960? I can't seem to get my xml right, and no matter what I send to the phone I keep getting parse errors. Thanks Shawn

I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the

I'm trying to get a basic connection setup between a nortel C10k and an asterisk box. All I want to be able to do is forward sip calls from the nortel switch to the asterisk box, and make calls from asterisk to the nortel.... ie: if someone dials my # on the PSTN, send it to asterisk and let me manage it, and if I dial 7xxx-xxxx drop the 7 and send it to the nortel switch I've done

Has anyone had any luck getting an asterisk box to talk to a Nortel C15K softswitch? I've been playing with it for several days and can't seem to pass calls either direction. I know that whike the Nortel says the C15K speaks SIP, it really speaks nortel's implementation of SIP, but I thought I could get it to at least pass simple calls back and forth to an asterisk box. Right now,

Is it possible for a Cisco 7960 phone with SIP firmware to connect to 2 different SIP servers @ the same time? I currently have an asterisk box @ home with several sip extensions and a Nortel C15k phoneswitch at work (not the pbx, the full phone switch). I can connect from the SIP phone to the Nortel phone switch, but cannot make asterisk talk to it at all (if anyone has any ideas on this one,

I have a situation where we have an asterisk box that is extending several Mitel PBX extensions to some cordless SIP phones (Cisco WIP310). Everything works great, except when the cordless phone walks out of range of one access point and into range of another (cisco 1100 series APs). I have another post to the list asking about how to speed up the handoff, and keep the call active while

I'm setting up an asterisk server to extend several extensions from a mitel pbx. I'd like to display the caller id that I receive from t he mitel pbx on the sip phone. The mitel PBX person has setup the PBX to send be callerid, but I don't see it. I've set chan_dahdi up with usecallerid=yes cidstart=ring cidsignalling=bell callerid = asreceived cid_rxgain = 0.0 Are there

I have 2 separate Asterisk servers that are both exibiting this problem. 1 has a 4 port FXO digium card, the other an 8 port. For some reason when the machine reboots, the dahdi drivers are not properly loaded. Then asterisk ends up starting without dahdi support. I've tried everything that I can think of, even to the point of running dahdi_cfg in the asterisk startup script before

I have a situation where we have an asterisk box that is extending several Mitel PBX extensions to some cordless SIP phones (Cisco WIP310). Everything works great, except when the cordless phone walks out of range of one access point and into range of another (cisco 1100 series APs). I've been able to get virtually seamless roaming between access points to work in the past with data but