similar to: Asterisk processing URI's

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

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 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

Typical user error, one user forwards his calls to another using CFwdAll on Cisco 7940, but the user receiving the call has done the reverse. -- Called 117 -- Got SIP response 302 "Moved Temporarily" back from 10.139.2.15 -- Now forwarding Local/117@sip-0eb7,2 to 'Local/114@sip' (thanks to SIP/117-df17) -- Executing Macro("Local/114@sip-35cf,2",

Hi, I got a problem of having to upgrade 35 Cisco 7960 phones from default firmware of 3.1 to 7.5. The problem I get is that when trying to upgrade I see on the tftplog that it can't seem to find the file (8 character issue). So I renamed the files to suit what is supposed to be in them. I am trying incremental upgrades from 3.1 -> 5.3 -> 7.5, with no luck. It goes to

Hi, I am having a weird issue with a Cisco 9971 phone. I managed to get most of it working, including the side car, however one of the issues is that there seems to be some sort of side tone / beep occurring roughly every 13 seconds or so, as if the phone is activated with call waiting. However none of this is activated and is still making the annoying beeping side tone. The phone does require

Hi, Just trying to figure out / solve some echo issues with one of our systems. The server is a Dell R710 running Asterisk 1.6.x and Dahdi 2.2.x The card in the server is a TE121P (with VPMADT032 echo canceller) We are experiencing some weird echo issues with our Cisco 79xx phones, only when we are dialing out via the PRI. The echo is like a side tone, where we can hear ourselves speaking. The

We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480. Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144). I contacted one of the developers and he suggested the passthrough of SDP attributes is not working correctly.

Hi, Anyone know any decent trunk provider in Canada that can offer multiple channel trunks (16channels) via IAX or Sip trunking? Having some pleasant experience with IAXTEL out of Denver, though they don't offer services into Canada. Please let me know S.

Hi, I was wondering if it is possible to setup with Asterisk a Cisco 7960 to use extension mobility / roaming. Meaning that a user logs into a phone and his profile moves with him / her. I have a network of ~75 Cisco 7960 phones, running SIP 7.5 distributed across 2 asterisk servers in 2 cities. I would like to enable the users to have the capability to move from one phone to another one

Does this help: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:1.2.3.4;transport=TCP SIP/2.0 Method: REGISTER Request-URI: sip:1.2.3.4;transport=TCP Request-URI Host Part: 1.2.3.4 [Resent Packet: False] Message Header Via: SIP/2.0/TCP 192.168.1.15:47053 ;branch=z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z-;rport;transport=TCP

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

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen

Hi, I was wondering if anyone had any problems with the Dell 1800 series servers, with TDM400 cards? I saw that digium seems to recommend against a lot of the dells. Please let me know Thanks Sascha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d8219499/attachment.htm

Hi, I am in need of ordering a new server here for our asterisk solution. Since the corporate standard is Dell we need to stick to a dell server. We used to deploy 2900III without any issues, however now they are almost not available any more and are looking at a new solution. Has anyone tried any of the new Dell R (series) servers with Asterisk, utilizing Digium PRI cards? The biggest issue I

Some of you may remember back in May the thread on using Asterisk as a voicemail server for a Cisco Callmanager system. My own Callmanager system is integrated into an Asterisk server for voicemail (and other things). Back in May I was using H323 for integration, but since I've upgraded to CCM 4.1 I have switched over to SIP. The integration with H323 required using Call forwarding to send

I have some Cisco 7905 phones with the SIP load 1.02.00(040406A). When the phone is off-hook but no call has been placed, or when the Do Not Disturb is activated, the phone returns a 302 "Moved Temporarily" message back to asterisk as follows: ----------- -- Executing Dial("SIP/5060-0811bb00", "SIP/9871234|20|Ttr") in new stack -- Called 9871234 -- Got SIP response

Hi, I am just trying to figure out how to setup a TDM04B card for incoming/outgoing calls. I have 4 lines, which are provided as a rotary trunk group, currently hooked into a Nortel system, which asterisk will replace. I have setup a Dell 1800 (Tower) system with the TDM04B card, which seems to work. The question is how do I set it up that all 4 lines are part of a trunk group, such that

Hello All! I have a strange problem with Cisco 7905. It is forwarding unanswered calls to VoiceMail even thought I have setup it not to. My ring timer on cisco 7905 is 60s, and my ForwardToVMDelay is 3000s. This means that call should never be forwarded to VM! This is true if I call from internal number then this happens on asterisk: -- SIP/104-6073 is ringing -- Nobody picked up in

Anyone have the 7970 xml config for sip yet? Aaron