similar to: Polycom IP600 - Bridge stops because we're zombie or need a soft hangup

Best R-users, Here’s a newbie question. I have tried to find an answer to this via help and the “ave(x,factor(),FUN=function(y) rank (z,tie=’first’)”-function, but without success. I have a dataframe (~8000 observations, registerdata) with four columns: id, dg1, dg2 and date(YYYY-MM-DD) of interest: id;dg1;dg2;date; 1;F28;;1997-11-04;

sip.conf.sample: ;register => 2345:password at sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf: [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no

Hey, Im running Asterisk 1.2.2 and im having problems with the audio when bridging calls between the zap interfaces and sip. zap to zap work fine, as do sip to sip (but asterisk isnt in the media stream, as it doesnt need to be) and terminating the call and playing a test message via either sip or zap work fine. Basically, the only time I see this problem is trying to bridge between sip and

Now this is interesting : du, doveadm quota get and maildirsize have three different values for this particular user : Max Quota is : 1G du : 883M (86%) maildirsize : 1048M (102%) doveadm : 34402? (32%) Trace ----- root at messagerie[10.10.10.19] ~ # cd /var/vmail/domain.tld/m.stefan/ root at messagerie[10.10.10.19] ~ # alias dush alias

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

What is the relationship between the peer definitions and the register command? In reviewing the sample sip.conf it seems that you can place the "sip_proxY" peer as the hostname. Is this correct? This question adds the the Broadvoice thread and where to place the dtmfmode variable. sip.conf --- (asterisk sample) -------------------------------- ;register =>

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

> Hello fellow asterisk people! > > I have Asterisk listening on port 5061 and SER on port 5060. > > Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. > > My problems are with SIP. I can make incoming calls from SIP to asterisk > and to any of the other networks, but when I try to make an outgoing call > from Asterisk to SER I see the following in

I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

Hi, I meet an amazing issue, when I convert a raw file to the oVirt environment using virt-v2v. All seems to work fine, my VM is composed of 9 disks, the processes finishes without any trouble, the files are well present in the right import volume, but only the first system disk appear un my oVirt Import VM. I've retried twice and had the same issue . Here the results of the conversion : [

I've tried this issue and could reproduce this issue: 1.Convert a guest with 2 disks to rhev. # virt-v2v -ic xen+ssh://10.66.106.64 -o rhev -os 10.66.90.115:/vol/v2v_auto/auto_export rhel6.6-i386-hvm -on 2disk-test -of qcow2 [ 0.0] Opening the source -i libvirt -ic xen+ssh://10.66.106.64 rhel6.6-i386-hvm [ 16.0] Creating an overlay to protect the source from being modified [ 47.0]

On Wed, Oct 08, 2014 at 08:11:16AM +0000, VONDRA Alain wrote: > Hi, > I meet an amazing issue, when I convert a raw file to the oVirt environment using virt-v2v. > All seems to work fine, my VM is composed of 9 disks, the processes > finishes without any trouble, the files are well present in the > right import volume, but only the first system disk appear un my > oVirt Import

Dear list users, I have a problem when joining an Active Directory domain. In this project we have one Main Dc in capital city and one read only dc in one remote city. We join to main DC succesfully. However, we can not join to local Replicate (rodc14). We are using this method for winbind / squid ntlm authentication purposes not a full samba server. ?nternet conection is not fast and we have

I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten => 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No

Hello, I have a very long (~50,000) sequence of repeating numbers. The first 100 are: [1] 0 0 0 0 0 0 0 0 0 0 0 429 [13] 429 429 429 429 429 429 429 858 858 858 858 858 [25] 858 1287 1287 1287 1287 1287 1716 2145 2145 2574 2574 3003 [37] 3003 3432 3432 3861 4290 4719 5148 5577 5577 6006 6006

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an