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Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

Every so often, and it seems that it happens only when a call is in progress, all 24 Zap channels get reset. All channels are opened and then timeout. This causes the in-progress calls to terminate. There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk although we do receive a fair amount of Blue Alarms. The Asterisk server is connected to a legacy PBX through a Digium

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

https://bugzilla.netfilter.org/show_bug.cgi?id=1344 Bug ID: 1344 Summary: Segmentation fault in nft add rule ip ipv4table ipv4chain-1 tcp sport { 12345-54321 } Product: nftables Version: unspecified Hardware: All OS: Ubuntu Status: NEW Severity: critical Priority: P5

Hi, Is it possible to break using if-condition during the recursive function? Here is a function which almost works. It is for inorder-tree-walk. iotw<-function(v,i,Stack,Indexes) # input: a vector and the first index (1), Stack=c(), Indexes=c(). { print(Indexes) # if (sum(i)==0) break # Doesn't work... if (is.na(v[i])==FALSE & is.null(unlist(v[i]))==FALSE) {Stack=c(i,Stack);

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

thank you for your response below the asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack --

Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn

I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060" Are you sure that "0033149xxxxxx" is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i

Hello, in the new Relase (1.4) i get a different (worser) result for read.table with as.is=T: it crash! Input file (t.txt, with a name, 5 character and a numeric column) Name short kind logable use save lag m "mo" "x" "n" "1" "n" 0 Ptp "PT" "l" "y" "m" "n" 0 R-Code

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings

I''ve recently re-installed an X4500 running Nevada b109 and have been experiencing ZFS lock ups regularly (perhaps once every 2-3 days). The machine is a backup server and receives hourly ZFS snapshots from another thumper - as such, the amount of zfs activity tends to be reasonably high. After about 48 - 72 hours, the file system seems to lock up and I''m unable to do anything

HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has