Re: [asterisk-users] [OT] How to use audio files with SIPp

Thank you very much George for replying. 2018-02-09 14:39 GMT+01:00 George Joseph : > > > On Fri, Feb 9, 2018 at 6:27 AM, Olivier wrote: > >> Hello, >> >> SIPp's PCAP play feature can replay pre-recorded audio stream towards >> destination (see [1]). >>

[asterisk-users] How to add SNMP support to packaged asterisk on Debian stretch

Hello, If I'm not mistaken SNMP support is missing in Debian Stretch packaged Asterisk while this support is present in either Jessie or Buster (looking at [1] or equivalent pages). Is it something that can be worked around or shall I fear a major obstacle when re-packaging my own asterisk

Re: [asterisk-users] [OT] How to use audio files with SIPp

On Fri, Feb 9, 2018 at 8:04 AM, Olivier wrote: > Thank you very much George for replying. > > 2018-02-09 14:39 GMT+01:00 George Joseph : > >> >> >> On Fri, Feb 9, 2018 at 6:27 AM, Olivier wrote: >> >>> Hello, >>> >>> SIPp's PCAP play

Re: [asterisk-users] [OT] How to use audio files with SIPp

On Fri, 9 Feb 2018, Olivier wrote: 3. How do you capture an RTP flux with thark or tcpdump ? Take a look at 'pcapsipdump.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST

[asterisk-users] Random, uncommanded blind transfers

Hi Guys I've got a situation where an incoming call originated from a trunk provider will ring in a call center and be answered by an agent. Usually within 15 seconds of the incoming call being answered, it will randomly blind-transfer to another extension in the same call center. It

[asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds

Re: [asterisk-users] AMD and fax detection

I think that there is a much easier way to detect if the far end is a fax. In your dialplan, include a section that uses the 'fax' extension. If the call jumps to that extension, then the far end is a fax machine. Regards; John V. -Original Message- From:

Re: [asterisk-users] AMD and fax detection

A quick test to a fax number doesn't seem to jump to the 'fax' extension. I checked faxdetect=both on chan_dahdi.conf. I seem to remember trying this some years back and finding that fax detect only worked for CNG tones, not CED tones? Neil Youngman

[asterisk-users] ast_iostream_close: SSL_shutdown() failed: Underlying BIO error: Bad file descriptor

Before I file a bug any ideas? Got valid certs, working fine for everything I need them from, but this seems to popup randomly in my logs. [Feb 9 12:43:07] ERROR[14968]: iostream.c:507 ast_iostream_close: SSL_shutdown() failed: error:0005:lib(0):func(0):DH lib, Underlying BIO error: Bad

[asterisk-users] AMD and fax detection

Is it feasible to enhance AMD to detect and report if the far end sends fax tones? I am guessing that, as it is using DSP to detect sounds and periods of silence, the same DSP can also report if a CED tone is sent. If it is feasible, is there a document describing the DSP interface that

[asterisk-users] [OT] How to use audio files with SIPp

Hello, SIPp's PCAP play feature can replay pre-recorded audio stream towards destination (see [1]). Doc mentions tcpdump and Wireshark as tools to record such RTP streams without further details. Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/ directory. Sample

Re: [asterisk-users] [OT] How to use audio files with SIPp

On Fri, Feb 9, 2018 at 6:27 AM, Olivier wrote: > Hello, > > SIPp's PCAP play feature can replay pre-recorded audio stream towards > destination (see [1]). > Doc mentions tcpdump and Wireshark as tools to record such RTP streams > without further details. > > > Looking at SIPp