search for: sip_general_custom

...checkip.dyndns.com | awk '{print $6}'| cut -d"<" -f1` is_ip $EXTERNIP if [ $? -ne 0 ] then logger -s "checksetexternip.sh: External IP address invalid or unavailable, exiting." exit 1 fi OLDEXTERNIP=`grep externip /etc/asterisk/sip_general_custom.conf | cut -d"=" -f2` if [ "$EXTERNIP" = "$OLDEXTERNIP" ] then logger -s "checksetexternip.sh: External IP address is the same, nothing to do exiting." exit 0 else logger -s "checksetexterni...

....c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down: WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image 16400 udptl t38 WARNING[3514] chan_sip.c: Failing due to no acceptable offer found sip_general_custom.conf contains t38pt_udptl=yes udptl.conf contains: [general] udptlstart=4000 udptlend=4999 T38FaxUdpEC = t38UDPRedundancy Asterisk version is 1.8.5.0 When I restart asterisk, everything is working good. Then, after some time, fax stop working. Do you have any idea what it could be? Thanks in ad...

I am running Trixbox 2.4 which has Asterisk 1.4.18-1 I have kind of followed: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 I added to sip_general_custom.conf ;NEEDED!!! t38pt_udptl = yes I did not add this to the actual SIP extension, as I assumed this being general it applies to all sip extensions, and doing a sip show peer ext# did indeed come up with t38pt_udptl = yes The fax is attached to a Grandstream 488, so I set it for fax mode: T.38...

...tension '70' priority 1 to parkedcalls (0xa8798b0) -- Reloading module 'res_phoneprov' (HTTP Phone Provisioning) == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/sip_general_additional.conf': == Found == Parsing '/etc/asterisk/sip_general_custom.conf': == Found == Parsing '/etc/asterisk/sip_nat.conf': == Found == Parsing '/etc/asterisk/sip_registrations_custom.conf': == Found == Parsing '/etc/asterisk/sip_registrations.conf': == Found == Parsing '/etc/asterisk/sip_custom.conf': == Found...

Does anyone know why? ThePBX*CLI> -- Executing [310-456-7890 at from-trunk:1] Set("SIP/202.101.202.101-b763ce60", "__FROM_DID=310-456-7890") in new stack -- Executing [310-456-7890 at from-trunk:2] ExecIf("SIP/202.101.202.101-b763ce60", "1 |Set|CALLERID(name)=310-456-0987") in new stack -- Executing [310-456-7890 at from-trunk:3]

Hello, What is the set of configuration that should be done in the Asterisk 1.0.8 using FreePBX that can allow a simple video call between two extensions? Thanks in advance. [http://www.ericsson.com/shared/images/Email_line.gif] JULIO ARAUJO TE ENGINEER MS Ericsson ITTE & Test Environment S?o Jose dos Campos, Brazil Phone +551239084121 SMS/MMS +551281150089 julio.araujo at

I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER