Archives : August-2016
Home » 2016 (paged 28)
I am having a problem with Fanvil phones (X3) when they make a call through DAHDI.Pure SIP calls flow normally but when a call goes through a DANDinterface to the PSTN we only get one way audio.This is Asterisk 13.10.0 (bundled pjsip) and Dahdi 2.1..
HiI have noticed that asterisk returns SIP 603 when the called party does not answer.My test setup is simple: two SIP phones (extensions: 100 and 111)registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and af..
Hellousing pjproject 2.5.5using asterisk-certified-13.8-cert1Compiled pjproject 2.5.5 with :./configure CFLAGS=-DNDEBUG -DPJ_HAS_IPV6=1 –prefix=/usr –libdir=/usr/lib64 –enable-shared –disable-video –disable-sound –disable-opencore-amrCompiled Aster..
The Asterisk Development Team has announced the release of Certified Asterisk 13.8-cert2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asteriskThe release of Certified Asterisk 13.8-cert2 resol..
list members, after programing of dialplan I have some messy Custom:hints which I can see in devstate list. I didnt find any possibility how to remove this hints from Asterisk and I want remove them. Can you help me with that, please? I tried sea..
What is needed to get DAHDI to start up correctly on CentOS 7 and systemd… I am using DAHDI-linux-complete 2.11.1I saw mention in my search that it has been fixed after 2.11.1 but cannot find what the fix is.Th..
All;What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension100 that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2..
Im trying to compile it with unbound but Im getting the following error:The UNBOUND installation appears to be missing or broken.Ubuntu 14.04.5 LTS \n \lroot@rtc:/usr/local/src/asterisk-14.0.0-beta1# dpkg -l | grep -i unboun iilibunbound-dev:amd64 1.4.22-1ubuntu4.14.04.2amd64sta..
Asterisk 13.11 rc1./configure LDFLAGS=-z muldefs –libdir=/usr/lib64–with-unixodbc=$(odbc_config –include-prefix)/ –disable-dev-mode–with-pjproject-bundledchecking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip… no checking for pjsip_tsx_create_u..
all,Just installed Asterisk 13 on CentOS 7 and have run into a problem.The Scenario is this:Asterisk is on the internet the Phone, a D40, is behind NATSo someone calls the number and Asterisk routes the call to the D40Everything works fine and the c..