search for: cisco7960

...P0S3-06-3-00 preferred_codec: g711ulaw #preferred_codec: g729a # Proxy Server proxy1_address: "10.6.0.223" proxy2_address: "" proxy3_address: "" proxy4_address: "" proxy5_address: "" proxy6_address: "" # Line 1 Settings line1_name: "Cisco7960" ; Line 1 Extension\User ID line1_displayname: "Cisco7960 Line1" ; Line 1 Display Name line1_shortname: "Line1" line1_authname: "Cisco7960"; Line 1 Registration Authentication line1_password: "Cisco7960" ; Line 1 Registration P...

I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the "Transfer" softkey does not show up - as a matter of fact only 2 softkeys show up (redial & something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are

Hello All, We are looking for a SIP provider teminating calls in India, Pakistan and Bengladesh. Any one knows a good one? Regards, Cesar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040824/4d5dd4b4/attachment.htm

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with SetCIDName. At least it's not displayed on my cisco7960 with chan_sccp any suggestions what I've could have done wrong ?

...e time in 24hour format date_format: D/M/Y ; format you would like the date in dial_template: dialplan SIP<MAC>.cnf: image_version: P0S3-8-12-00 line1_name: 55 line1_authname: 55 line1_shortname: 55 ; displayed on the phones softkey line1_password: 12345655 line1_displayname: "Lukasz Cisco7960"; the caller id proxy1_port: 5060 proxy1_address: 192.168.1.109 # Phone Label (Text desired to be displayed in upper right corner) phone_label: "Castle " ; add a space at the end, looks neater phone_password: "cisco" ; Limited to 31 characters (Default - cisco) user_info:...

I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had...

HI all: I have spent a large amount of time configuring/installing phones connected to Asterisk. Halfway through the process I discovered that my Cisco7960 with 2 7914 expansions was not supported in the SIP protocol. After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of configuring SCCP to properly work with Asterisk. So far I have gotten the phone to dial and receive calls from the other participating SIP 7.2 phones on the LAN....

Hi, I have succeed using Asterisk on VMWare on an Athlon@1GB with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium...

...mic' in sip.conf. I want calls to these extensions to be routable even before the device registers. I understand that is what defaultip= is supposed to do, but it doesn't work. I get a busy tone when dialing the extension until the phone reregisters. Here is what the entry looks like for a Cisco7960: [cisco] type=friend username=cisco secret=supersecret host=dynamic defaultip=192.168.0.55 canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away context=local callerid="Cisco Phone" <2010> mailb...

....168.5.228 -- SCCP: >> Got message AlarmMessage -- SCCP: Alarm Message: Severity: Informational (2), 25: Name=SEP000D288E2257 Load=4.0(03.00) Last=CM-closed-TCP [2049/1234] -- SCCP: >> Got message RegisterMessage -- SEP000D288E2257: is registering, Instance: 1, Type: Cisco7960 (7), Version: 5 -- SEP000D288E2257: Allocating device to session (21) 192.168.5.163 -- SEP000D288E2257: Building button template Cisco7960(7), user config 7920 -- SEP000D288E2257: Phone available lines 6 -- SEP000D288E2257: Auto logging into 880 -- SCCP: Looking for line 88...

I am currently testing a new Asterisk installation. The server has a T100P connected to a PRI, and about 50 Polycom IP600 phones connected via the local network. Every couple hours, Asterisk randomly stops responding to all calls, both incoming on the PRI and calls from the SIP phones. I'm not sure how or where to start debugging it. When Asterisk stops working I can still connect to

...ia_port: "32766" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "WaZaB-SIP" ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: EST # Phone prompt/password for telnet/console session phone_prompt: "Cisco7960" ; Telnet/Console Prompt phone_password: "abc" ; Telnet/Console Password # SIP Configuration Generic File (stop) SIPDefault.cnf # Image Version image_version: "P0S3-06-0-00" # Proxy Server proxy1_address: "10.1...

Hi *, does someone has a directory that works with the Cisco 7960 and astdb or mysql/ldap? Regards, Andreas _________________________________________________________________ Gaming galore at http://xtramsn.co.nz/gaming !

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Shaun Ewing > Sent: Tuesday, November 04, 2003 7:15 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Anyone using * in a live > production environment? > [...] > The additional features are nice too.

...with numbered extensions for demonstration purposes. What is the syntax to associate a extension with SIP phone? Does the Dial application have a SIP specific entry for example: Dial,SIP/SIPphone/s|15 When I call from one extension to another I get "User is on the phone". We also have Cisco7960s to test. Currently Have X-Lite setup. Can log into the server and can execute the demo features. No CO line interface. Just experimenting with IP. SIP.conf configured voicemail.conf configured. extensions.conf configured. sip.conf [user] callerid="User Name&quot...

...I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers with no telephone-event in OK. Only Sipura3000 "manages" to get answer with telephone-event in OK and that's why asterisk detects dtmfs. I try to experiment with dtmfmode in sip.conf with no results. S...

This might have been talked about before, but I'm posting anyhow. I've got down to testing Asterisk yesterday, and I couldn't help but compare it with Skype (a Windoze only product, yet, but extremely efficient for some reason). Skype has almost unperceptible delay (LAN), while there is almost half a second of delay regardless of the codec on Asterisk. An even if we were to