similar to: Softkeys wrong with chan_skinny

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

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, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

Hi, I'm trying to replace function call with call to wrapper(function_name, num_args, ...), where varargs hold args of original call. Function* launch = Function::Create( TypeBuilder<int(const char*, int, ...), false>::get(context), GlobalValue::ExternalLinkage, "kernelgen_launch_", m2); { CallInst* call = dyn_cast<CallInst>(cast<Value>(I)); if

Hi Dimitry, This makes sense if you think about it from the perspective that the string you want passing must be passed at runtime, and so can't use a const char * from compile time. You need to make the string visible in the compiled image, and use that as the argument. A string is an array of 8-bit integers, so you need to create a ConstantArray. Value *v = ConstantArray::get(Context,

Dear useRs, Some time ago I queried the list as to an efficient way of building a function which acts as ls() but with a different default for all.names: http://tolstoy.newcastle.edu.au/R/e6/help/09/03/7588.html I have struck upon a solution which so far has performed admirably. In particular, it uses ls() and not its explicit source code, so only has a dependency on its name and the name of

if i go off hook and dial 666 from an internal sipura spa-x000 (at extn 141), it rings straight through to extn 666. using the same dialplan, from a cisco 7960 with 7.1 sip code (at extn 142), i have to go off hook hit NewCall punch 142 (or any valid extn in the dialplan) hit Dial then dial 666 wtf? sip.conf for crisco [fiji] callerid="crisco" <142>

Hello all, I'm in final testing stages and preparing training for a new Asterisk rollout. I'm replacing a Cisco Call Manager system, and re-flashing the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call, I use the "Confrn" softkey to invite other participants. I can add one other participant endpoint into the conference, but no more. I know I can (and

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, ? 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

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 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

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

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

I'm a day-old R newbie (but a war-weary S veteran), with couple of first-day questions: In R 0.61, this code fails. Ops.test <- function(e1,e2) { e1 <- NextMethod(.Generic) e1 } x <- 4 class(x) <- "test" y <- x < 3 The error message is "Error in NextMethod(.Generic) : negative length vectors are not allowed.". I assume it is a bug.

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Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, Asterisk is running strictly VoIP over the network and using CCM as the trunk. Calls from the SIP phones connected to Asterisk work fine. They can call both external numbers and any Cisco extensions attached to CCM. Calls from CCM to Asterisk fail without any notification in Asterisk (and I DID have this working at one

I am looking for a simple way to forward calls unconditionally with Asterisk. We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary, when she is out, she needs to be able to forward unconditionally to her mobile or collegue. I am trying to keep it as simple as possible, we use Cisco 7940's, they have a call forward option, when she

does anyone know howto set the softkeys of an Cisco 7960 running on an asterisk server via chan_sccp ? Mit besten Gr??en Dirk Rieger