Re: [asterisk-users] Voice "broken" during calls
On Saturday 13 June 2020 at 22:30:28, Luca Bertoncello wrote: > 1) I have an Android phone, using the integrated Android VoIP-subsystem, > connected to my Asterisk at home, over LTE or other network *outside my > home network*. > I called my mother using this method... The quality was excellent > 2) I have a Thomson ST2022 connected to my Asterisk over Ethernet > (cabled network). If I call for example my mother or my parents in law, > the conversation is "broken", eg: both partner can hear little > "interruption", about 1/10 seconds in the conversation... I would like to see a much simpler one-for-one comparison: only change one thing at a time, and see what the difference is. So: I suggest you try *two* independent *pairs* of tests: 1a. Using your Android phone, connect using your home wireless network (I assume you have a wireless network, if not then skip to test 2) to your home Asterisk server, make a phone call to some external number, check the call quality. 1b. Using your Thomson phone, connected using your home cabled network to your home Asterisk server, make a phone call to the same external number and check the call quality. 2a. Using your Android phone, connect from outside your home wireless network over LTE to your home Asterisk server and make a phone call to the same number again (you'll need someone with a bit of patience and understanding on the other end of this number ...) Check the call quality. 2b. Take your Thomson telephone to some other location with Internet access, let it register to your home Asterisk server, and them make a call to the same number yet again. I'm sure you can get the Thomson to connect to Asterisk via some external network, since you say you can do this from your Android phone. Again, check the call quality. Then, does the call quality stay the same for both phones (good for Android, bad for Thomson), or does it stay the same for both connections (good for Android and Thomson from external, bad for Android and Thomson from internal)? PS; Just for fun, what happens if you let your Android phone connect via LTE to your home Asterisk server and you dial your (home, cabled) Thomson phone from it? What's the call quality like then? In regard to: On Saturday 13 June 2020 at 18:25:32, Luca Bertoncello wrote: > 2) where can I change these settings? sip.conf Look for lines such as disallow=all allow=ulaw allow=alaw allow=h263 They may be in the [general] section, or they may be in the client (Android / Thomson) specific sections. Regards, Antony. -- The lottery is a tax for people who can't do maths. Please reply to the list; please *don't* CC me. -- _ -- 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] Voice "broken" during calls
Am 13.06.2020 um 22:09 schrieb Antony Stone: Hi Antony > You are *assuming* that it's the codec causing the difference. Well, I really don't know what I can think, now... > We don't know that. > > Let me get this clear, to make sure I understand (differences emphasised): > > 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, > to your Asterisk server, over your home *wireless network*, to place a call > to > some external number, you have a conversation and *the quality is excellent*. > > 2. You use your *Thomson ST2022*, which is also registered by SIP, to your > home Asterisk server, over your home *cabled* network, to place a call to > some > (the same???) external number, you have a conversation and the quality is > *not > excellent*. > > > Is that an accurate summary of your situation? Not really... 1) I have an Android phone, using the integrated Android VoIP-subsystem, connected to my Asterisk at home, over LTE or other network *outside my home network*. Today I called my mother using this method (I was in the home network of my parents in law, about 20km von my home network, so definitly *not* in my wireless...). The quality was excellent and it was confirmed from my father in law, too... 2) I have a Thomson ST2022 connected to my Asterisk over Ethernet (cabled network). If I call for example my mother or my parents in law, the conversation is "broken", eg: both partner can hear little "interruption", about 1/10 seconds in the conversation... This is the situation... I tried to connect the Thomson ST2022 directly to the server of Deutsche Telekom via VoIP (excluding the Asterisk, but of couse using NAT, since the phone does not have a public IP but just an IP in my internal network) and then I called my father in law. Same problem... :( I didn't get my Android phone connected to the server of Deutsche Telekom to check how it works *outside my home network*... Not sure why it doesn't work... Some other information: 1) Asterisk runs on a Linux-Box (on a BananaPI) with Debian 10. Asterisk was installed from Debian repositories. 2) The Linux-Box is directly connected to the Internet (no NAT) with a DSL-Modem and PPPoE. Public IPv4 and IPv6 addresses are configured in a network interface of the Linux-Box. 3) I use iptables+tc to manage a traffic shaping, privileging the VoIP connection. If you want, I have no problem to send the traffic-shaping-script to the list. 4) The DSL connection has a speed of 50Mbps down and 10Mbps up, and I really think, it should be enough... 5) The phones are connected with Gbps-Ethernet to the Linux-Box. 6) On my Asterisk I configured a second VoIP-Provider (MessageNet, in Italy), but just to *receive* calls. My contract with MessageNet does not allow me the call someone using this connection. If someone calls my number by MessageNet, I have the same problem I have with Deutsche Telekom, altought not so strong, eg. the "interruptions" are not so frequent as by calls via Deutsche Telekom... Btw: by MessageNet I must use *gsm* as Codec, otherwise a connection will be extablished, but no Voice can be heared... I really appreciate any idea. Of course, it could be possible that there is a problem on Telekom-side, but it does not explain why I have the same problems, altought not often as by Telekom, by MessageNet, too... Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Voice "broken" during calls
On Saturday 13 June 2020 at 18:26:53, Luca Bertoncello wrote: > Am 13.06.2020 um 18:06 schrieb Michael Keuter: > > So the call used Alaw as Codec. > > Yes, so seems it to be... > It should has the better quality... But the calls done using my mobile > phone in VoIP with the Asterisk have better quality as the calls done > using the normal VoIP-telefon... You are *assuming* that it's the codec causing the difference. We don't know that. Let me get this clear, to make sure I understand (differences emphasised): 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, to your Asterisk server, over your home *wireless network*, to place a call to some external number, you have a conversation and *the quality is excellent*. 2. You use your *Thomson ST2022*, which is also registered by SIP, to your home Asterisk server, over your home *cabled* network, to place a call to some (the same???) external number, you have a conversation and the quality is *not excellent*. Is that an accurate summary of your situation? Antony. -- Just when you think you're done, a cat floats by with buttered toast strapped to its back. - Steve Krug, "Don't make me think" Please reply to the list; please *don't* CC me. -- _ -- 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] Voice "broken" during calls
Am 13.06.2020 um 18:06 schrieb Michael Keuter: > So the call used Alaw as Codec. Yes, so seems it to be... It should has the better quality... But the calls done using my mobile phone in VoIP with the Asterisk have better quality as the calls done using the normal VoIP-telefon... I'm really puzzled... Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Voice "broken" during calls
Am 13.06.2020 um 18:20 schrieb Antony Stone: Hi >> bpi*CLI> sip show peer 0049177xxx >> Codecs : >> (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin| >> slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|t >> estlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk >> |silk|silk) > > That strikes me as somewhat unlikely. Too much things, isn't it? >> bpi*CLI> sip show peer 0049351xxx >> Codecs : (alaw|ulaw|ilbc|g729|g723|gsm) > > That looks a little more standard. The questions are: 1) why the mobile phone, with "too many things" has a better quality 2) where can I change these settings? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Voice "broken" during calls
On Saturday 13 June 2020 at 17:23:14, Luca Bertoncello wrote: > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > > Try "sip show peer " for a phone. > bpi*CLI> sip show peer 0049177xxx > Codecs : > (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin| > slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|t > estlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk > |silk|silk) That strikes me as somewhat unlikely. > bpi*CLI> sip show peer 0049351xxx > Codecs : (alaw|ulaw|ilbc|g729|g723|gsm) That looks a little more standard. Regards, Antony. -- I just got a new mobile phone, and I called it Titanic. It's already syncing. Please reply to the list; please *don't* CC me. -- _ -- 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] Voice "broken" during calls
On Saturday 13 June 2020 at 18:06:23, Michael Keuter wrote: > So the call used Alaw as Codec. ...which should be excellent quality. PS: Michael: thanks for the tips regarding "sip show channels" and "sip show channel " - I was aware of these for some details, but hadn't realised they showed the codecs as well. Very useful :) Antony. -- Your work is both good and original. Unfortunately the parts that are good aren't original, and the parts that are original aren't good. - Samuel Johnson Please reply to the list; please *don't* CC me. -- _ -- 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] Voice "broken" during calls
So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello : > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer " for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxx > > > > > * Name : 0049177xxx > > > Description : > > > Secret : > > > MD5Secret: > > > Remote Secret: > > > Context : default > > > Record On feature : automon > > > Record Off feature : automon > > > Subscr.Cont. : > > > Language : de > > > Tonezone : > AMA flags: Unknown > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup: 1 > Pickupgroup : 1 > Named Callgr : > Nam. Pickupgr: > MOH Suggest : > Mailbox : > VM Extension : asterisk > LastMsgsSent : 0/0 > Call limit : 2147483647 > Max forwards : 0 > Dynamic : Yes > Callerid : "0049177xxx" <> > MaxCallBR: 384 kbps > Expire : -1 > Insecure : no > Force rport : Yes > Symmetric RTP: Yes > ACL : No > DirectMedACL : No > T.38 support : Yes > T.38 EC mode : FEC > T.38 MaxDtgrm: 4294967295 > DirectMedia : No > PromiscRedir : No > User=Phone : No > Video Support: No > Text Support : No > Ign SDP ver : No > Trust RPID : No > Send RPID: Yes > Path support : No > Path : N/A > TrustIDOutbnd: Legacy > Subscriptions: Yes > Overlap dial : No > DTMFmode : rfc2833 > Timer T1 : 500 > Timer B : 32000 > ToHost : > Addr->IP : (null) > Defaddr->IP : (null) > Prim.Transp. : UDP > Allowed.Trsp : UDP > Def. Username: > SIP Options : (none) > Codecs : > (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk) > Auto-Framing : No > Status : UNKNOWN > Useragent: > Reg. Contact : > Qualify Freq : 6 ms > Keepalive: 0 ms > Sess-Timers : Refuse > Sess-Refresh : uac > Sess-Expires : 1800 secs > Min-Sess : 90 secs > RTP Engine : asterisk > Parkinglot : > Use Reason : No > Encryption : No > > VoIP-phone (Thomson ST2022): > bpi*CLI> sip show peer 0049351xxx > > > > > * Name : 0049351xxx > > > Description : > > > Secret : > > > MD5Secret: > > > Remote Secret: > > > Context : default > > > Record On feature : automon > > > Record Off feature : automon > Subscr.Cont. : > Language : de > Tonezone : > AMA flags: Unknown > Transfer mode: open > CallingPres : Presentation Allowed, Not Screened > Callgroup: 1 > Pickupgroup : 1 > Named Callgr : > Nam. Pickupgr: > MOH Suggest : > Mailbox : > VM Extension : asterisk > LastMsgsSent : 0/0 > Call limit : 2147483647 > Max forwards : 0 > Dynamic : Yes > Callerid : "0049351xxx" <> > MaxCallBR: 384 kbps > Expire : 3111 > Insecure : no > Force rport : Yes > Symmetric RTP: Yes > ACL : Yes > DirectMedACL : No > T.38 support : Yes > T.38 EC mode : FEC > T.38 MaxDtgrm: 4294967295 > DirectMedia : No > PromiscRedir : No > User=Phone : No > Video Support: No > Text Support : No > Ign SDP ver : No > Trust RPID : No > Send RPID: Yes > Path support : No > Path : N/A > TrustIDOutbnd: Legacy > Subscriptions: Yes > Overlap dial : No > DTMFmode : rfc2833 > Timer T1 : 500 > Timer B : 32000 > ToHost : > Addr->IP : 192.168.200.10:25572 > Defaddr->IP : (null) > Prim.Transp. : UDP > Allowed.Trsp : UDP > Def. Username: 0049351xxx > SIP Options : (none) > Codecs : (alaw|ulaw|ilbc|g729|g723|gsm) > Auto-Framing : No > Status : OK (17 ms) > Useragent: THOMSON ST2022 hw2 fw3.56 00-26-44-31-10-23 > Reg. Contact : sip:0049351xxx@192.168.200.10:25572;user=phone > Qualify Freq : 6 ms > Keepalive: 0 ms > Sess-Timers : Refuse > Sess-Refresh : uac > Sess-Expires : 1800 secs > Min-Sess : 90 secs > RTP Engine : asterisk > Parkinglot : > Use Reason : No > Encryption : No > > >> Then "sip show channels" during an existing call. > > Call from normal phone: > bpi*CLI> sip show channels > Peer User/ANR Call ID Format Hold >Last MessageExpiry Peer > 192.168.200.10 0049351xxx 9eff88f7-c0a801 (alaw) No >Rx: ACK0049351xxx > 217.0.27.53 03501xxx 453efbcb7a04f33 (alaw) No >Tx: ACKpbxluca > 2 active SIP dialogs > > Call from mobile phone (via VoIP registered in Asterisk): > > bpi*CLI> sip show channels > Peer User/ANR Call ID Format Hold >Last MessageExpiry Peer > 192.168.10.120049177xxx 11b86bd612b71a
Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?
On Sat, 13 Jun 2020, Jonathan H wrote: I need to ensure that a MusicOnHold stream is only running when there's a caller on hold and listening.To do that, I need to rewrite and reload the moh.conf file when the caller hangs up IF there are no other callers (ie there's just 1 active call as the caller hangs up), and then rewrite and reload again when there's a new caller. How about ARA to configure MOH and then just update the database. -- 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] Voice "broken" during calls
Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer " for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxx * Name : 0049177xxx Description : Secret : MD5Secret: Remote Secret: Context : default Record On feature : automon Record Off feature : automon Subscr.Cont. : Language : de Tonezone : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "0049177xxx" <> MaxCallBR: 384 kbps Expire : -1 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: Yes Path support : No Path : N/A TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : (null) Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk) Auto-Framing : No Status : UNKNOWN Useragent: Reg. Contact : Qualify Freq : 6 ms Keepalive: 0 ms Sess-Timers : Refuse Sess-Refresh : uac Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No VoIP-phone (Thomson ST2022): bpi*CLI> sip show peer 0049351xxx * Name : 0049351xxx Description : Secret : MD5Secret: Remote Secret: Context : default Record On feature : automon Record Off feature : automon Subscr.Cont. : Language : de Tonezone : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "0049351xxx" <> MaxCallBR: 384 kbps Expire : 3111 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : Yes DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: Yes Path support : No Path : N/A TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.200.10:25572 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 0049351xxx SIP Options : (none) Codecs : (alaw|ulaw|ilbc|g729|g723|gsm) Auto-Framing : No Status : OK (17 ms) Useragent: THOMSON ST2022 hw2 fw3.56 00-26-44-31-10-23 Reg. Contact : sip:0049351xxx@192.168.200.10:25572;user=phone Qualify Freq : 6 ms Keepalive: 0 ms Sess-Timers : Refuse Sess-Refresh : uac Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No > Then "sip show channels" during an existing call. Call from normal phone: bpi*CLI> sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer 192.168.200.10 0049351xxx 9eff88f7-c0a801 (alaw) No Rx: ACK0049351xxx 217.0.27.53 03501xxx 453efbcb7a04f33 (alaw) No Tx: ACKpbxluca 2 active SIP dialogs Call from mobile phone (via VoIP registered in Asterisk): bpi*CLI> sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer 192.168.10.120049177xxx 11b86bd612b71ae (alaw) No Rx: INVITE 0049177xxx 217.0.27.53 00493501xxx 5647efe41d746b4 (alaw) No Tx: INVITE pbxluca 2 active SIP dialogs > And "sip show channel " for more info. Call from normal phone: bpi*CLI> sip show channel 9eff88f7-c0a80101-0-22c911@192.168.200.10 * SIP Call Curr. trans. direction: In
Re: [asterisk-users] Voice "broken" during calls
> Am 13.06.2020 um 13:36 schrieb Luca Bertoncello : > > Am 13.06.2020 09:30, schrieb Luca Bertoncello: > > Hi again (again) > > I noticed right now another strange detail... > I made a call using my mobile phone (connected to the Asterisk). The quality > was top... > Maybe is the problem in a codec used from our phones at homes? > Could someone suggest me how to check the codec used by my mobile phone and > the codec used by the phones at home? > > Thanks > Luca Bertoncello > (lucab...@lucabert.de) Try "sip show peer " for a phone. Then "sip show channels" during an existing call. And "sip show channel " for more info. Michael http://www.mksolutions.info -- _ -- 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] Voice "broken" during calls
On Saturday 13 June 2020 at 13:36:00, Luca Bertoncello wrote: > Am 13.06.2020 09:30, schrieb Luca Bertoncello: > > Hi again (again) > > I noticed right now another strange detail... > I made a call using my mobile phone (connected to the Asterisk). What does that mean? You're making a mobile phone call over the GSM network to your Asterisk server, or you're using a soft phone application on your smartphone, which is registered by SIP to your Asterisk server? Also, where did you make the call *to* ? > The quality was top... > Maybe is the problem in a codec used from our phones at homes? Didn't we already discuss this last year? http://lists.digium.com/pipermail/asterisk-users/2019-December/294446.html > Could someone suggest me how to check the codec used by my mobile phone > and the codec used by the phones at home? Look at the verbose log file and search for "transcoding". Also, do a SIP packet trace at the start of the call and see which codecs are announced by each side and then what gets agreed on (I don't think this gets logged by Asterisk, so you need to look at the SIP negotiation itself). Antony. -- I'm not impossible, just highly implausible. Please reply to the list; please *don't* CC me. -- _ -- 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] Voice "broken" during calls
Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Any api (agi/ari/ami) equivalent of "core show calls"?
I'm parsing ` sudo asterisk -rx "core show calls" | grep active | head -c 1 ` as an external call from within the Asterisk dialplan then passing it to agi, but this seems really hacky and ugly. However, I cannot find any ARI/AGI/AMI function (or global variable I can get with agi) which shows me this. Any ideas?!? In case it helps and you're wondering why... I need to ensure that a MusicOnHold stream is only running when there's a caller on hold and listening. To do that, I need to rewrite and reload the moh.conf file when the caller hangs up IF there are no other callers (ie there's just 1 active call as the caller hangs up), and then rewrite and reload again when there's a new caller. -- _ -- 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] Voice "broken" during calls
Am 13.06.2020 um 08:28 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to understand what my > partner sayd... > > It can NOT be a problem of other downloads/uploads, since in that moment > there were no ones... Hi again! Just a detail: I tried an internal call (from my phone, to my wife's phone) and it works wonderful, no broken, no delay, top quality. So the problem _MUST_ be in the settings of the communication with Deutsche Telekom and MessageNet (the providers I used). The settings for Deutsche Telekom are: [pbxluca] type=peer defaultuser=-0001 secret= dtmfmode=rfc2833 host=tel.t-online.de context=luca_incoming outboundproxy=tel.t-online.de port=5060 fromuser=0351xxx fromdomain=tel.t-online.de usereqphone=yes canreinvite=yes insecure=port,invite nat=force_rport,comedia qualify=yes qualifyfreq=600 disallow=all allow=alaw allow=ulaw and the settings for MessageNet are: [messagenet] type=peer defaultuser= secret= dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5060 fromuser= fromdomain=sip.messagenet.it usereqphone=yes canreinvite=yes insecure=invite qualify=yes qualifyfreq=60 disallow=all allow=alaw allow=ulaw allow=gsm Any idea? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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
