similar to: configuring a Cisco 7961 so that different line appearances register to different SIP proxy addresses

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 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, ? 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

Anyone have the 7970 xml config for sip yet? Aaron

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

I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 phones in a paging group. I have all the phones set up with an extra line that auto answers the dial from my paging extension when the primary line is not in use. All of these are operating correctly however the 7961/7970s all ring once and then auto answer so the person paging all the phones has the first part of his

How does asterisk process URI's that get sent to it? I am having a issue with a Cisco phone, where 99% works except the call forwarding. The phone issues a X-cisco-serviceuri-cfwdall which can be seen when running a sip debug on the peer directly. However the system tries to lookup the request as a extension, aka X-cisco-serviceuri-cfwdall-<extension> And of course can't find this

Hi all, I succeed in injecting audio into one channel by mean of ChanSpy, but this audio cannot be listened correctly. I am using softphones for Android and iPhone and there are so many cuts so that they cannot understand what is said in the audio file. Is this because the total RTP bandwidth is too much?. Which codec and format should I use for the files played with Playback application?

On 05/12/14 16:46, Olli Heiskanen wrote: > INVITE that Asterisk (at port 5070) receives: > PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 > INVITE sip:660 at testers.com > <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP >

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

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.

Hello, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

Well, I thought it worked, but it actually doesn't--I am able to get the caller pick up the phone, but for some reason, I cannot hear anything on either side no matter who does the calling. Again, my two SIP phones are on the local 192.168.1.0/24 network (do not go over the Internet) and the Asterisk server is located in the same network (not accessed over the Internet). Any help is

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (