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 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 pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to >>> 10.1.6.18:2006 >>> >>> 1. How can you "forge" IPs and/or ports of a pcap file ? >>> >> >> You don't have to. sipp only takes the rtp payload from the packets in >> the pcap then just sends the datagrams to the remote in the scenario. >> > > That is exactly what I'm after ! > > Before diving into this, can I ask which SIPp version and feature are we > talking about here ? > We're on 3.5. > > Since I posted my question, I've read this [1] thread mentionning a new > WAV file playing capability but this feature required SIPp 3.4 and above. > On Debian Stetch I'm playing with, packaged SIPp is 3.2. > That's pretty old. I'd recommend compiling from source yourself. It's very easy to build. > > > [1] https://stackoverflow.com/questions/20122607/playing- > audio-file-using-sipp/20123193 > > >> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Check out the new Asterisk community forum at: >>> https://community.asterisk.org/ >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> >> >> -- >> George Joseph >> Digium, Inc. | Software Developer >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US >> Check us out at: www.digium.com & www.asterisk.org >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Tech Support [aster...@voipbusiness.us] Sent: 09 February 2018 14:53 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: 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: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neil Youngman Sent: Friday, February 09, 2018 09:10 AM To: asterisk-users@lists.digium.com Subject: [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 someone who is not familiar with DSP can use to get started? Neil Youngman Neil Youngman Developer Wirefast Limited Wirefast provides secure corporate messaging services. See our messaging solutions at http://www.wirefast.com/ Please consider the environment. Does this email or attachment need to be printed? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Any views or opinions are solely those of the author and do not necessarily represent those of Wirefast Limited Email transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. Wirefast Limited is registered in England & Wales Company number: 03865860 Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ Wirefast definitions of classification can be found here: www.wirefast.com/classifications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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]). >> 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 pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to >> 10.1.6.18:2006 >> >> 1. How can you "forge" IPs and/or ports of a pcap file ? >> > > You don't have to. sipp only takes the rtp payload from the packets in > the pcap then just sends the datagrams to the remote in the scenario. > That is exactly what I'm after ! Before diving into this, can I ask which SIPp version and feature are we talking about here ? Since I posted my question, I've read this [1] thread mentionning a new WAV file playing capability but this feature required SIPp 3.4 and above. On Debian Stetch I'm playing with, packaged SIPp is 3.2. [1] https://stackoverflow.com/questions/20122607/playing-audio-file-using-sipp/20123193 > >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > > > -- > George Joseph > Digium, Inc. | Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 package for Stretch ? Comments ? Suggestions ? Best regards [1] https://packages.debian.org/buster/asterisk-modules -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neil Youngman Sent: Friday, February 09, 2018 09:10 AM To: asterisk-users@lists.digium.com Subject: [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 someone who is not familiar with DSP can use to get started? Neil Youngman Neil Youngman Developer Wirefast Limited Wirefast provides secure corporate messaging services. See our messaging solutions at http://www.wirefast.com/ Please consider the environment. Does this email or attachment need to be printed? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Any views or opinions are solely those of the author and do not necessarily represent those of Wirefast Limited Email transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. Wirefast Limited is registered in England & Wales Company number: 03865860 Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ Wirefast definitions of classification can be found here: www.wirefast.com/classifications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 after 3082 has started speaking to the outside caller - 3082's call goes dead in their handset. - The outside caller goes back into the queue, hears queue MOH and gets answered by another person in the office as if they are dialing in all over again. - 3082's phone starts ringing again after they hang up in puzzlement and if they then pick up they speak to another person who is trying to make an OUTGOING call in their call center. This is for a medium sized call center which (along with 17 other centers in the same country) run the same dialplan on Asterisk 1.8.32.3 - only happens at this location. Literally 100 000+ calls are handled across these 18 centers every day, only about 10 or 20 at this one center (with carbon-copy dialplan and SIP phone hardware types - Yealink T-21Ps - as at every other branch) keeps disconnecting people picked up and transferred from the incoming queue from the person transferred to, and then connects them and the transferree to other phones - the caller as if he is phoning in AGAIN into the incoming queue, the transferred-to person to someone else who is trying to dial out once the transferred-to person has hung up after losing the incoming caller. Anybody ever encountered something similar? The same dialplan on the same Ast version runs fine in 17 other locations, some with ten or twenty times more traffic and none of these issues. No errors or strangeness apparent in the CLI, verbose log, DTMF log... Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 is as if Asterisk is mis-reading some noise in-band as DTMF and doing the transfer, or the caller is emitting DTMF to transfer. An examination of CEL records for the relevant call shows the transfer taking place, but no party (the caller or callee) tried to transfer. The relevant queue application already has only small "t" in the parameter list (e. g. only let the called user - e. g. my agent inside - transfer the incoming call), thereby preventing the caller maybe emitting DTMF down the line and transferring at their behest. Yet the random, uncommanded blind transfers still take place, always withing about 15 seconds after call initiation in answering an incoming SIP call. We're using YeaLink SIP phones - T21P - got 17 branches, this only happens at 3 of them at random times. Same hardware and Asterisk versions (1.8.32.3) at all of them, same dialplans. Operating over about several years, about 10 000 000 incoming calls handled so far, and this only started happening now... dialplans are untouched for months. Any ideas, pointers, anybody encountered this before? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 someone who is not familiar with DSP can use to get started? Neil Youngman Neil Youngman Developer Wirefast Limited Wirefast provides secure corporate messaging services. See our messaging solutions at http://www.wirefast.com/ Please consider the environment. Does this email or attachment need to be printed? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Any views or opinions are solely those of the author and do not necessarily represent those of Wirefast Limited Email transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. Wirefast Limited is registered in England & Wales Company number: 03865860 Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ Wirefast definitions of classification can be found here: www.wirefast.com/classifications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 3.2 source archive, I found PCAP samples in a pcap/ > directory. > Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to > 10.1.6.18:2006 > > 1. How can you "forge" IPs and/or ports of a pcap file ? > You don't have to. sipp only takes the rtp payload from the packets in the pcap then just sends the datagrams to the remote in the scenario. > > 2. When generating simultaneous calls from one source device to a single > target device, do you need to have specific PCAP files (one specific for > each call) with specific source port ? > Nope. See above. > > 3. How do you capture an RTP flux with thark or tcpdump ? > This is a little tricky. tcpdump isn't much help if you don't know the ports. With tshark though you can create a "read" filter that can capture only RTP packets but it's very expensive. Usually, I just capture all UDP packets between the hosts in question with tcpdump, then use the Wireshark gui to filter the rtp stream. Then you can just export those packets. > > Best regards > > [1] http://sipp.sourceforge.net/doc/reference.html#PCAP+Play > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to 10.1.6.18:2006 1. How can you "forge" IPs and/or ports of a pcap file ? 2. When generating simultaneous calls from one source device to a single target device, do you need to have specific PCAP files (one specific for each call) with specific source port ? 3. How do you capture an RTP flux with thark or tcpdump ? Best regards [1] http://sipp.sourceforge.net/doc/reference.html#PCAP+Play -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 file descriptor Something I'm doing? Asterisk 15.2.0 on Ubuntu 17.10 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
