Archives : July-2016
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Many people are reporting the same issue, so it is not my imagination. Asterisk 13 above 13.1 is useless for anybody who relies on res_odbc.so. As you know, after that version, the dropped the complexity of Pooling onto unix_odbc itself. Not..
Sporadically we get 1 way audio when one party is outside our firewall.The caller is on NAT, and it works fine most of the time. Caller can hear the called party, same thing going the other direction. Caller can hear called party.Asterisk 13.9 on Deb..
Hello.Anybody in the list is using this IP phone?Regards,Marcelo H. Terres IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/mar..
I have two questions:1- Mailbox on the Asterisk Voicemail Server are created automatically?2- Is there any support on the code to put the voice records on a Cassandra NoSQL database?BRJoaquin This email is confidential and may be subject to privile..
im trying replace CDR with CELreasons:- minimize Stasis listeners (CDR)- CEL, CDR produces similar data- own logic of CDR meaning like calldate,src,dst,direction,.. dst is always first connected point in PBX – real user or IVR/queue etc., numbers ..
everyone.Im trying to compile Asterisk with the smsq utility on Ubuntu 16.04 LTS, and while most things are compiling fine, smsq fails with the following output:root@test25:/usr/src/asterisk-certified-13.1-cert7/utils# make smsq[CC] smsq.c -> smsq.o[..
Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL.What would be the proper way to do this for Asterisk 13 and PJSIP? — Telecomunicaciones Abier..
How is it possible to use Dial application to force out-bond call use specified channel number in one E1 or specified CIC (SS7) ? Regard..
I am still a little confused about how to activate MWI with PJSIP on Asterisk 13.9.1.I use realtime for configuration.So far I have tried setting both the mailboxes field on ps_endpoints and the mailboxes field in ps_aors but I cannot get the indica..
Im using Asterisk 13.9.1 (compiled from source) in Ubuntu 14.04.4.Im testing webrtc with jscommunicator (www.jscommunicator.org), and Im using wss. My problem is as follows:If I setdtls_cipher=ALLI can get audio with Firefox 47.0.1, but with Chrome 52.0.2743..