similar to: Asterisk - Cisco 7960 - NAT

What happens when you do "stop now" like the error states? Sean -----Original Message----- From: Ryan Parlee [mailto:listbox@jesca.com] Sent: Saturday, April 03, 2004 9:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unabled to exit console No matter what I try, Asterisk won't let me out of the console. If I CTRL+C, of course, the process will terminate. I

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 -

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual Band Analoog FXO) working with Asterisk. I had a working FXO configuration to a analog port of a small home 1/4 ISDN pbx. I used this same configuration to connect a GSM Gateway that is supposed to be connected to the external(FXO) analog port of a pbx. I can get my configuration to dial the mobile number via the gateway, but

Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment:

Running those versions of code, my 7960 will not register with Asterisk. The same 7960 is authenticating against another * server on line 2 just fine though - with the same settings in sip.conf. On the failing * server I am just getting 401 unauthorized errors on the console. From the phone's shell I get that t is registering, but not authenticated .1. from show reg. Any ideas would be

Hello, all. In section 4.3.10 of the Asterisk Handbook, there is an example of an LCR/Failover Trunking scenario. I've tried it, and it works, as long as I fail over from something else to ZAP, but I can't get it to "hunt" to the other context if the zapata channel (or group) is used first. Can anyone help? Here is my extensions.conf, and the error message I get.

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

I have some strange behavior happening with call flow when analog line errors are encountered. This may be due to the way that the X100P detects "busy" signals, or it may be something in the software. Could someone with more in-depth knowledge make a comment on the items below? My dialing logic says "dial local area code numbers out of the analog line, and if the analog line

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck.

I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both

I've been pulling out my hair all day on this one. If anyone can help, I'd really appreciate it. :-( I've got an Aastra 9133i (with the latest firmware version) and a Cisco 7960 sitting behind a NAT device on my LAN. The Asterisk server is hosted offsite and has a public IP address. I've set up port-forwarding on the firewall for both phones to tunnel the SIP messages initiated

Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing "55 <phone icon with x>" so it looks like the phone is not registered. The phone and the asterisk are in the same local

Thanks to everyone for your quick responses to this question. I'm very excited about the Asterisk project, and the growing community seems to be very active these days. Hopefully when the time comes for our county's transition to VoIP we may be able to go for an Asterisk-based solution. -- Tony Kava Network Administrator Pottawattamie County, Iowa -----Original Message----- From:

Hi, I can't get dial-out working. I'm trying to call 523936. Is there something wrong with my setup here? Could someone please give me a few pointers? Regards, Norman Zhang [fwd-out] exten => _8.,1,SetCallerID(${FWDUSERID}) exten => _8.,2,SetCIDName(${FWDUSERNAME}) exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70) exten => _8.,4,Macro(fastbusy) exten => _8.,5,Hangup *CLI>

Started a new thread as my problem is somewhat different than the OP. Seems his somewhat different problem doesn't work as advertised either. Eric Wieling wrote: > I don't know what the problem is, but this is what I use and it works on my analog FXO port. > exten => _9NXXNXXXXXX,1,Dial(${PSTN}/w${EXTEN:1}) So, I modified slightly to fit my dialplan: exten =>

I'm trying to clarify my understand of gosub, macros and AEL. My understanding is that macros using the Macro() application, which is defined in extensions.conf by: [macro-foo] ... and called in extensions.conf with exten => _9NXXNXXXXXX.,n,Macro(fastbusy) is deprecated in favour of Gosub(). True so far? But then there are "macro"s defined in extensions.ael: macro foo() {

Hi, I wish to connect several ATA186 Phones to each other, to iconnecthere and to the PSTN using asterisk. Please tell the appropriate settings for firewall (ports to open etc.) sip.conf and extensions.conf(part relevant to iconnect). Also I would be glad to get a working example of your ATA186 configuration. I tried searching the mailing lists and several sites but did not find an answer.