similar to: Registering Cisco 7942G IP phone with Asterisk!.

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

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

Hi, Does anyone know how to define speeddials in XML for the 7970 sip firmware?. I've played with the SEP<mac>.cnf.xml file that was posted previously but can't find a way to do it. I can define them on the phone usually (seems a bit buggy) but if the phone reboots they get lost from the config. Does anyone know a way to do this? I'm guessing it could be within the

- How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is <line button="4"> <featureID>9</featureID> ... For speeddial is <line button="5"> <featureID>2</featureID> <featureLabel>341</featureLabel> <speedDialNumber>341</speedDialNumber> </line>

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:

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

hi all If I try to register a cisco 7945 phone (firmware sip v 9.2) to a asterisk server 1.8.5 I set in the xml file SEPmac.cnf.xml <callManager> <processNodeName>IP_ADDRESS_ASTERISK</processNodeName> ... and in the line 1 settings <proxy>USECALLMANAGER</proxy> it works good . Now I need to use the phone with 2

Hi, ? The current SCCP image on the 7942 phone is :SCCP42.9-0-2SR1S. We are trying to convert/upgrade the phone to SIP version of the firmware i.e : cmterm-7942_7962-sip.9-0-3 (Firmware is downloaded from the cisco support site). We have unzipped and placed all the files in the /tftp (root directory) of tftp server. Following files are also placed in the tftp directory. ? The Upgradation /

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 (

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

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

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

Hello All, I've been trying to get BLF working with Asterisk 1.6-LatestBeta, and My Cisco 7970 (Latest SIP Firmware). Has anyone successfully completed this? I got the patch to merge in from http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones With a bit of hackery to that code, it successfully compiled chan_sip for me. So, With 1.6 we have TCP Enabled on Asterisk,

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.

Hi I am using xen 3.0.4 in sles10 sp1 . Now I want it to be upgraded to xen 3.1. my machine is x86/32 bit. Before xen 3.1 I have to install xen 2.6.18.....(current version is xen 2.6.16....). > When I upgraded the kernel to 2.6.18-4-xen-686, my dom0 system cannot > startup and I get this error during the kernel loading: > > PANIC on CPU0: > Could not setup DOM0 guest OS xen