We have a FreePBX-12 / Asterisk-12 setup that supports about 24
extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x)
and is up-to-date. Registrations require very long random passwords and registrable devices are further restricted by netblock filters. We have the usual firewall and fail2ban intrusion prevention and detection set-ups in place.
Our connection to the rest of the world is via PSTN.
We do our own DNS, both forward and reverse. We have NAPTR and…
I have a client that wants a phone system that will play sounds from a sleep machine. I tried using all different formats (GSM, WAV, WV49, MP3 etc.). Over SIP it was OK however with the PSTN it broke up from time to time. I assume this has to do with the fact that the PSTN is limited to 8khz. Is there something I am missing here or is this simply a limitation of the PSTN?
I have 2 Asterisk systems and a unique scenario where I need to play a particular tone on Asterisk1 and identify the same tone on Asterisk2. Following is my call flow, Asterisk1(Plays audiofile1,Wait for 2 Sec,Plays a tone,Plays audiofile2) -> PSTN -> 3rd Party CONFERENCE SYSTEM <- PSTN <- Asterisk2(Record audiofile1,Wait for a tone,Record audiofile2). A few points to keep in mind, (1)I can not send DTMF tones as Conference system suppresses it. (2)There is no other way to pass information from Asterisk1 to Asterisk2 (3)Asterisk2 doesn't know the length of audiofile1,audiofile2. (files are less than 200 Sec in duration)
I've got a small install with Asterisk 11. This box is connected to PSTN through a SIP trunk.
I need to add a cellular phone as a remote agent of an existing queue.
At the moment, this queue is configured according a ringall strategy and busy agent can be dialed.
My plan is : 1- to create a remote-agent context and insert a Wait statement into it so that local agents would get dialed ahead of remote agents. 2- to avoid dialing remote agents already on call (with local phones) by hanging up calls in remote-agent context when conditions are met. 3- use…
On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side.
When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering >0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:portnumber then not until about 6 seconds later I see this >0xhexnumber -- Probation passed - setting RTP source address to 192.168.1.11:diffportnumber and immediately hear audio
what appears to be an issue is that the RTP link(audio)…
On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), users are complaining for bad audio.
My setup is: PSTN <--E1/PRI ---> Asterisk <--- E1/PRI---> Siemens HiPath <---E1/PRI ---> PSTN
asterisk -rx "dahdi show version" DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC
asterisk -rx "pri show version" libpri version: 1.4.12
A quick glance at Asterisk logs shows lines like this: [2014-01-09 17:19:34] NOTICE: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 [2014-01-09 17:19:35] NOTICE: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 2 [2014-01-09 17:19:49] NOTICE: chan_dahdi.c:3099 my_handle_dchan_exception: PRI got event: HDLC Abort (6)…
I'm using Asterisk 11.7, connected to PSTN using SIP trunk.
I'm looking for a way to get data from INVITE's SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the call originates from.
I have googled for a solution and found this patch: https://issues.asterisk.org/jira/browse/ASTERISK-14510 which looks like exactly what i need, but, unfortunately looks like it was abandoned or forgotten.
The patch is against older version of chan_sip and i don't know how to apply it against the current version. I'm not enough familiar with chan_sip internals.
When I call an specific number on the PSTN, the provider who holds the destination number give back an specified sound just after admitting their incoming calls. Is there a way to allow Asterisk to compare sounds received to decide what is the Telco answering the call?
I'm planning to do it to select the right provider to route further calls at least cost.
In my country there are no public or accessible information on ported out numbers, so it is a way to discriminate what are the destination's Telco and build a database.
Thanks in advance!
Elder D. Arohuanca Lima - Peru
I always thought that Energy Savings mode existed with ISDN Basic Rate Interface in Point -to-multi-Point but it didn't with Point-to-point.
Is this correct ? Have you ever met a public PSTN switch configured to "cut" B ISDN channels in Point-to-point.signalling ?