similar to: Cisco 7970 running SIP question

Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question Aaron Yes it

Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running

Anyone have the 7970 xml config for sip yet? Aaron

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello; I am facing a problem to let Cisco IP Phone 7942G register on Asterisk. The firmware has been downloaded from the TFTP successfully and currently I am running this load SIP42.9-3-1SR2-1S* I feel that there is a problem in the SEPMAC.cnf.xml but really I do not know which one to be used exactly. Basically, there is some effect that appears on the Phone (for example, it is appearing the

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear "scratches" and "plops". It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to

I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great !

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute

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

I'm getting Number but when I look at the CDR database. I do see the name -----Original Message----- From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you

Hi everyone, I have a question about faxing. I am running Asterisk SVN-trunk-r7498 on a ubuntu server and everything is going fine. I have a tdm40b, a tdm04b and an avm fritz!card pci plugged on it. Here is what I'd like to do: Receiving fax via the ISDN avm fritz!card, sending the fax via email (for the moment I know I can do that) but also routing the fax trough my old fax machine which

I've got this in extensions.conf: [macro-stdexten] exten => s,1,Dial(${ARG2},30,p) exten => 6015555555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+15045555555) Where the real numbers have been replaced with 5555555. What I'm trying to do is ring my cell phone in addition to the local extensions. Funny thing is the cell phone rings

Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for

Hello, I'm currently developing a simple asterisk application using SFS (Skype For SIP) which tries to call to an outbound number, play a message and read DMTF digits. My first approach used the Manager to originate calls and then called an agi script to deal with the rest. Anyway, this ended up being not so clear because the call did not start on the Originate extension that it was supposed

I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following: exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70) There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the

Hi, I'm seeing a very strange error when dealing with Diversions. If a call setup to a number comes to an Asterisk server, that server sends a request to a third proxy, that proxy sends the call back with a Diversion flag, Asterisk complains about the host not existing (and the host is the number). Here's the output from the Asterisk CLI with SIP debugging enabled: <--- SIP read from