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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

...tion failed to produce executable FAIL: gfortran.dg/char_transpose_1.f90 -Os (internal compiler error) FAIL: gfortran.dg/char_transpose_1.f90 -Os (test for excess errors) WARNING: gfortran.dg/char_transpose_1.f90 -Os compilation failed to produce executable FAIL: gfortran.dg/fmt_p_1.f90 -O0 execution test FAIL: gfortran.dg/fmt_p_1.f90 -O1 execution test FAIL: gfortran.dg/fmt_p_1.f90 -O2 execution test FAIL: gfortran.dg/fmt_p_1.f90 -O3 -fomit-frame-pointer execution test FAIL: gfortran.dg/fmt_p_1.f90 -O3 -fomit-frame-pointer -funroll-loops execution test FAIL: gfortran.dg/fmt_p_1.f90 -O...

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

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 just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

...unroll-all-loops -finline-functions (test for excess errors) FAIL: gfortran.dg/namelist_13.f90 -O3 -g (test for excess errors) FAIL: gfortran.dg/namelist_13.f90 -Os (test for excess errors) FAIL: gfortran.dg/namelist_14.f90 -O0 (test for excess errors) FAIL: gfortran.dg/namelist_14.f90 -O0 execution test FAIL: gfortran.dg/namelist_14.f90 -O1 (test for excess errors) FAIL: gfortran.dg/namelist_14.f90 -O1 execution test FAIL: gfortran.dg/namelist_14.f90 -O2 (test for excess errors) FAIL: gfortran.dg/namelist_14.f90 -O2 execution test FAIL: gfortran.dg/namelist_14.f90 -O3 -fomit-frame-po...

Hi, I''m trying to run Xen 3.0 with a file-backed VBD on a opteron system. Here''s my Xen config file (xenconf): # -*- mode: python; -*- kernel = "/boot/vmlinuz-xen" ramdisk = "/boot/initrd-xen" memory = 1024 name = "Dom2" vif = [ '''' ] disk = [ ''file:/xendisk/vm3disk,sda1,w'' ] root = "/dev/sda1

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

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

...ilding llvm 2.3 and llvm-gcc4.2-2.3 on Mac OS X 10.5, I am seeing the following failures remaining in the gcc 4.2.1 gfortran testsuite... LAST_UPDATED: Native configuration is i686-apple-darwin9 === gfortran tests === Running target unix FAIL: gfortran.dg/actual_array_constructor_1.f90 -O1 execution test FAIL: gfortran.dg/actual_array_constructor_1.f90 -O2 execution test FAIL: gfortran.dg/actual_array_constructor_1.f90 -O3 -fomit-frame-pointer execution test FAIL: gfortran.dg/actual_array_constructor_1.f90 -O3 -fomit-frame-pointer -funroll-loops execution test FAIL: gfortran.dg/actual_ar...

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 >