Re: [asterisk-users] asterisk18.6.0 with old chan_sip and microsoft teams
On Fri, Aug 13, 2021 at 2:21 PM Jerry Geis wrote: > Hi, > > I had a different thread going about about no audio with asterisk - I > thought it was due to two network cards - but I dont think so any more. > The endpoint is microsoft teams - and I think that might be the issue. > > Does Asterisk 18.6.0 work with Microsoft teams ? my issue is no audio. > > I have done a tcpdump -i eth0 and I see the audio traffic going back on > that network - so I presume I am good - What can I do so teams sees the > audio ? > > Thanks, > > Jerry > > The configuration is this: "initiating a call from Teams, that call gets routed through an Audiocodes 2600 SBC. That 2600 SBC then passes the call over to the Marion Audiocodes Mediant 1000. " So teams -> 2600 SBC -> Mediant 1000 -> Asterisk 18.6.0 - Call does come in - looks normal but they report no audio heard back at source. Thanks, Jerry -- _ -- 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] asterisk18.6.0 with old chan_sip and microsoft teams
On Friday 13 August 2021 at 20:21:23, Jerry Geis wrote: > Hi, > > I had a different thread going about about no audio with asterisk - I > thought it was due to two network cards - but I dont think so any more. > The endpoint is microsoft teams - and I think that might be the issue. Please show how you have got Asterisk working with MS Teams in the first place. As far as I understand it, this hasn't been possible without placing an SBC such as Kamailio in the middle, but if it is possible, that would be interesting. > Does Asterisk 18.6.0 work with Microsoft teams ? my issue is no audio. I'm slighly impressed you appear to have the initial registration working. > I have done a tcpdump -i eth0 and I see the audio traffic going back on > that network - so I presume I am good - What can I do so teams sees the > audio ? What does something like sngrep or tshark/wireshark tell you is going on with the SDP negotiations? Antony. -- Atheism is a non-prophet-making organisation. 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
[asterisk-users] asterisk18.6.0 with old chan_sip and microsoft teams
Hi, I had a different thread going about about no audio with asterisk - I thought it was due to two network cards - but I dont think so any more. The endpoint is microsoft teams - and I think that might be the issue. Does Asterisk 18.6.0 work with Microsoft teams ? my issue is no audio. I have done a tcpdump -i eth0 and I see the audio traffic going back on that network - so I presume I am good - What can I do so teams sees the audio ? Thanks, Jerry -- _ -- 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] problems with natted phones
Hello, it triggered again. Even disabling RTSp ALG did not help. My fantasy ends here. It agains seems to be reboot triggered on asterisk side. Not every one. But there was surely one before it was last working. Reboot of the router on the phone side fixes the problem. Any other suggestions? This is where you use sngrep or tcpdump to look at whats actually happening on the asterisk box. sngrep is focussed on sip dialogs and is probably easier than tcpdump when you are just interested in sip If you use sngrep on the asterisk server sip port you will see the SIP packet flows for registration and call setups. You can check the addresses given out for rtp to respond to and the codecs. Is an address incorrect? Is a code incorrect? You will see in the session description protocol what codecs the client is requesting and what the replies are asterisk works well around the world with many nat scenarios so I imagine its either config or firewall. A firewall with ALGs is often problematic but your log suggests a lack of negotiation of agreed codecs. Good luck, you will learn some interesting things. Thanks Marek 2021-07-26 9:31 GMT+02:00, Marek Greško : I currently disabled also RTSP ALG and rebooted the router. Fixed for now. I do not know for how long. Marek 2021-07-26 8:54 GMT+02:00, Marek Greško : Hmm, back to original problem. My happines was premature. Today one of the phones have no audio again. I see packets from lan segment again. I double checked the router configuration. SIP ALG is disabled. There are also another ALGs present: NAT ALG RTSP ALG PPTP ALG IPSEC ALG Which of them are neede to be disabled? As of my observations these problems are triggered by reboots on asterisk side. How could this be related? (I may be wrong.) Thanks Marek 2021-07-23 14:54 GMT+02:00, Marek Greško : I achieved a partial success adding --use-compact-form option. Marek 2021-07-23 13:47 GMT+02:00, Marek Greško : Hello, your suggestion to turn off SIP ALG on provider's router was probably correct. no problem until now. Thank you very much. I just found out another issue. I had a pjsue client in that network which called specific number when turned on. It was working perfectly with the old provider with working SIP ALG. But now with this provider and SIP ALG disabled I am not able to make the call using pjsua client. My pjsua config looks like this: --id sip:ext@asterisk.domain --registrar sip:asterisk.domain --proxy sip:asterisk.domain --outbound sip:asterisk.domain --realm * --username username --password password --null-audio --no-tcp --max-calls=1 --no-vad The pjsua client successfully registers but is unable to call. I see the following: IP address change detected for account 1 (localip:5060-->nattedip:newport). Updating registration (using method 4) Temporary failure in sending Request msg INVITE/cseq=, will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) What could be the problem? How can I convince pjsue to work correctly behind nat? Thanks Marek 2021-07-10 11:08 GMT+02:00, Marek Greško : Hello, I just disabled. Currently it is working. I shloud give it some time to confirm the problem has gone. Maybe one month would be enough to confirm. Thanks Marek 2021-07-09 20:11 GMT+02:00, Abdenasser Ghomri : Yes just disable the SIP ALG and see if it helps, Thanks. Best Regards, On Fri, Jul 9, 2021, 09:10 Antony Stone < antony.st...@asterisk.open.source.it> wrote: On Friday 09 July 2021 at 08:47:46, Marek Greško wrote: > Hello, > > yes SIP ALG are anbled on the router. Should I disable? In my opinion, always. Antony. -- I don't know, maybe if we all waited then cosmic rays would write all our software for us. Of course it might take a while. - Ron Minnich, Los Alamos National Laboratory 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 -- _ -- 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] asterisk 18 logger
Hi
I saw the example of asterisk18 log as follows:
logger.conf.sample
If the verbose level is not specified, it
; will log verbose messages following the current level of the root console.
so my asterisk CLI> core show settings is Root console verbosity: 3
my logger.conf change settings to 1 verbose => verbose(1)
The current question is, why can I see the dial extensions process?
[08-13 16:14:12.701] VERBOSE[271148][C-0007] pbx.c: Executing [s@xx:1]
Set("PJSIP/incoming-no-encrypt-000e", "__PJV1=read,") in new stack
[08-13 16:14:12.701] VERBOSE[271148][C-0007] pbx.c: Executing [s@xx:2]
Set("PJSIP/incoming-no-encrypt-000e",
"__PJV2=b(macro-pjsipvariable^predial_b_leg^1(PJSIP/incoming-no-encrypt-000e))B(macro-pjsipvariable^predial_a_leg^1)")
in new stack
[08-13 16:14:12.702] VERBOSE[271148][C-0007] pbx.c: Executing [s@xx:3]
Set("PJSIP/incoming-no-encrypt-000e", "__CHANNELpeerip=") in new stack
[08-13 16:14:12.702] VERBOSE[271148][C-0007] pbx.c: Executing [s@xx:4]
Set("PJSIP/incoming-no-encrypt-000e", "__CHANNELrecvip=") in new stack
--
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