I want to offer SIP phone user a custom fax-to-email feature. Here is how I would describe this feature:- for every SIP phone,a custom email address is defined- when a SIP phone answers an incoming call (from a trunk or another SIPendpoint), it dete..
Im trying to diagnose potential causes of an issue with MixMonitor in 1.4.43(I doubt its very version-specific). I wont have hands on the kit until the end of the week, but I have listened to some recordings. It doesnt happen on every call – only sometimes.Basical..
all,I need some help understand the values of the CHANNEL function, e.g.My main problem in understand is that a CHANNEL has two nodes (sender and receiver), while a typical setup includes at least 3 nodes:SIP phone – Asterisk – SIP Provider ( -> e..
On one of our locations, I am having issues with one-way audio when I call several phones with SIP/Phone_A&SIP/Phone_B&SIP/Phone_C. When I call the phones individually, they work fine, so its not a volume setting on the phone. Also this setup has wor..
1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. E.g. how about a software that extracts all calls from the /var/log/asterisk/full (I have permanently enabled verbose 10 and sip deb..
Kevin, I have two asterisk boxes with the same issues. Box 1: asterisk ver 126.96.36.199 Box 2: Asterisk 188.8.131.52 setup: CDMA PhoneCDMA Media Gateway WCMAsterisk voice mail The calls are SIP Based.DTMF collection is when the user is entering a password ..
I use an SBC to protect my pool of asterisk servers and as trunking endpoint with SIP Telcos. Now Im trying to implement SIP phone registration to be delegated through the SBC, as a proxy. It doesnt work. It just works when I dont use realtime pe..
I would like to test the following COLP use case : Alice and Bob are both using a SIP phone registered on a Asterisk 10 server. Alice dials Bobs extension. While Bobs phone is ringing, Asterisk updates Alice phone screen with Bobs name, so that a..