similar to: chan_capi upgrade

I am trying to make an outgoing call using an iptel account using Asterisk. I have followed a how-to for asterisk and iptel found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER I am getting the following error message: Got SIP response 403 "That is ugly -- use From=id next time (OB)" back from 195.37.77.101 I'm not quite sure what that means. Does

Hello asterisk-users, I recently set up an asterisk server using Debian Sarge. I also added an ISDN card (AVM FritzCard PCI) to the machine. I built amd installed a new kernel (2.6.15.1) with modular support for the CAPI stuff and also integrated the FritzCard driver available from AVM. capiinfo succeeds in correctly showing the ISDN card. When talking from one PC to another via SIP, everything

On Jul 15, 2015, at 12:51 PM, William Dunlap wrote: > I think rapply() was changed to act like lapply() in this respect. > When I looked at the source of the difference, it was that typeof() returned 'language' in 3.2.1, while it returned 'list' in the earlier version of R. The first check in rapply's code in both version was: if (typeof(object) != "list")

David, If you are referring to the solution that would be: rapply(list(test), eval, envir = fenv) I thought I explained in the question that the above code does not work. It does not throw an error, but the behavior is no different (at least in the output or result). Using the above code still results in the x object not being stored in fenv on 3.1.2. Dayne On Wed, Jul 15, 2015 at 4:40 PM,

I am trying to enable dial-by-email by using LDAPget to query an Active Directory server. I've got it retrieving the phone number fine. Unforunately, the numbers stored in active directory are either in the format: (xxx) xxx-xxxx or xxx-xxx-xxxx. Is there any way to parse characters out of the dialed phone number so that I only end up with digits (remove spaces, parenthesis and dashes)?

Hi all, i have asterisk 1.0.9 with an Eicon Diva 4bri and chan_capi-cm-0.6. I have 2 NTBAs (one with did and one without). When using the one without did, i am able to place outgoing and incoming calls. When i use the NTBAs with did i have a layer 2 error. Anyone an idea? -- Executing Dial("SIP/2004-9634", "CAPI/g1/43XXXXXX") in new stack > data = g1/43XXXXXX

Hello, These are error messages I get when I try to call a number over CAPI channel. -- Executing SetCallerID("SIP/xlite1-3b80", "0") in new stack -- Executing Dial("SIP/xlite1-3b80", "CAPI/hfcpci/b17") in new stack > data = hfcpci/b17 > capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00,

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

Hello, Anyone knows how can I do this in a cleaner way? mynumber = 1001 as.numeric(unlist(strsplit(as.character(mynumber),""))) [1] 1 0 0 1 Thanks in advance, Gustavo

Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the "@" in front of the outbound msn in the dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten

I'm getting a strange error on one of the two controllers on an AVM C2 card under chan_capi-cm-0.6.3. I have two ISDN controllers defined, both in the same group, both connections are UK ISDN2e Point to Point: On the third outbound call (both of the first two calls are handled by the second controller "ISDN2",) I get this error : chan_capi.c conf_error 0x2001 PLCI=0x301

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML

I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Please help _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft.

Hi, the german telco Colt Telekom has assigned the phone number block 56830-xxx to one of our customers. In the diaplan we have setup extensions like the following ones: exten => 56830910,1,Answer() exten => 56830910,2,Dial(SIP/bduerring,10,tr) exten => 56830910,3,VoiceMail,u20 exten => 56830910,4,hangup exten => 56830910,103,VoiceMail,b20 exten => 56830910,104,hangup exten

Hello All, I am having an odd problem with Armin's chan-capi_cm on builds higher than 7955. It would seem that this happens on anything higher than 7955. What is happening is the isdn is ringing, then asterisk does a goto-if and just hangs. Asterisk itself is ok, but the isdn then rings out or busys out on the other side. Outgoing works fine, this only seems to effect incoming. I

We intend to use Asterisk with a very large dialplan (with a lot of functionality for 3000+ users). Each user will be able to change several of his parameters in the dialplan, so we will be forced to reload the diaplan constantly. Has anybody else any previous experience with a similar installation? There are some things that we'd like to know, if anybody can help us. These are: - Is

Hello List, because this is my first post to this list, i'd like to introduce myself. My name is Thorsten M?ller, 26 years old and live near Frankfurt/Main in germany. Okay, now to the reason for this posting: I just installed my first asterisk server (Debian 2.2 Kernel 2.4.18-686) with AVM ISDN Fritz PCI card (passive). I followed the configuration on http://voip-info.org and the

Nik, This definitely helps! Please check your dial command. You've got "Dial(Zap/0/mynumber)" and I think you might possibly want it to be something like this: Dial(Zap/1/mynumber) or Dial(Zap/g0/mynumber) I don't recall there being a zap channel zero, but it is common to have a group zero. I would recommend trying Zap channel 1 - Dial(Zap/1/mynumber) - before trying the

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",