Re: [asterisk-users] Local calls not possible when Internet connection down
Hello, I confirm server and phones are on the same subnet and the phones are able to resolve local domain also when internet connection os down. It seems to be the asterisk bug I referenced before. There seems to be some bolcking resolver in it. I do not use database related to asterisk. This should be related to the srv record resolving. It seems quite random time to trigger the issue. When inspecting logs after internet problems started the issue appeared in one hour and several minutes. After restart of the asterisk it reappeared in less than half an hour. When trying to reproduce I was not able to reproduce for one hour and a half. So I decided to configure srv_lookups=no. I hope the issue is workarounded now. But I think asterisk should be fixed. It should successfully start when the VoIP providers sip server is not reachable, should recover after it becomes available. And should work locally when it stops to be responding. The tweak of creating /etc/hosts entry for the sip server and disabling srv lookups should not be needed. I hope sometimes theese issues will be addressed. Marek On Wednesday, November 8th, 2023 at 15:53, John Harragin wrote: > Are the phones and the server in the same subnet? You might making note of > the IPs and just simply try pinging everything with the uplink disconnected. > Also, if you are using domain names for registration, it is possible a dns > server must be reachable. > > If you are using database for any of your call processing, an unreachable dns > server can also be the cause of trouble. For some reason, even if you are > using IP addressing, Mysql will try to resolve a connection and can hang > (there is a mysql parameter to not resolve addresses). > > On Wed, Nov 8, 2023 at 8:46 AM Marek Greško > wrote: > >> Hello, >> >> it did not seem the call hung. It seemed it never started. There was no >> dialplan execution on the asterisk side. It looked like phones were >> unregistered. Same shows the log posted previously. >> >> Marek >> >> Sent with Proton Mail secure email. >> >> --- Original Message --- >> On Wednesday, November 8th, 2023 at 1:21, John Harragin >> wrote: >> >>> Marek, >>> >>> See if calls hang in the system if you encounter another outage >>> core show channels >>> >>> ...if so, >>> core set verbose 3 >>> and see what instructions subsequent calls hang on. >>> >>> >>> >>> On Mon, Nov 6, 2023 at 4:44 PM Marek Greško marek.gre...@protonmail.com >>> wrote: >>> >>> > Hello, >>> > >>> > sure I have local DNS server and public resolving should not be needed >>> > for phone registrations. Running pjsip show endpojnt show the endpoints >>> > as not in use. >>> > >>> > When looking into logs I see only res_pjsip_outbound_registration.c: No >>> > response >>> > received from sip provider. Nothing else. >>> > >>> > In phone log I see: >>> > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), >>> > lid=0, par=0, par2=(nil)) >>> > >>> > The phone is Cisco SPA525G2. >>> > >>> > Thanks. >>> > >>> > Marek >>> > >>> > --- Original Message --- >>> > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp jc...@sangoma.com >>> > wrote: >>> > >>> > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško marek.gre...@protonmail.com >>> > wrote: >>> > >>> > > It looks like all phones get unregistered, but I am not aware of the >>> > > cause. Why are get not registered when there is a connectivity between >>> > > them and asterisk? >>> > >>> > Are the REGISTER requests reaching Asterisk (do they show up in a packet >>> > capture, do they show up in "pjsip set logger on")? It needs to be >>> > further isolated. How are the phones configured to reach Asterisk? If >>> > using a hostname, are they still able to resolve it? >>> > >>> > -- >>> > Joshua C. Colp >>> > Asterisk Project Lead >>> > Sangoma Technologies >>> > Check us out at www.sangoma.com and www.asterisk.org >>> > >>> > -- >>> > _ >>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> > >>> > Ch
Re: [asterisk-users] Local calls not possible when Internet connection down
Hello, it did not seem the call hung. It seemed it never started. There was no dialplan execution on the asterisk side. It looked like phones were unregistered. Same shows the log posted previously. Marek Sent with Proton Mail secure email. --- Original Message --- On Wednesday, November 8th, 2023 at 1:21, John Harragin wrote: > Marek, > > See if calls hang in the system if you encounter another outage > core show channels > > ...if so, > core set verbose 3 > and see what instructions subsequent calls hang on. > > > > On Mon, Nov 6, 2023 at 4:44 PM Marek Greško marek.gre...@protonmail.com wrote: > > > Hello, > > > > sure I have local DNS server and public resolving should not be needed for > > phone registrations. Running pjsip show endpojnt show the endpoints as not > > in use. > > > > When looking into logs I see only res_pjsip_outbound_registration.c: No > > response > > received from sip provider. Nothing else. > > > > In phone log I see: > > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > > lid=0, par=0, par2=(nil)) > > > > The phone is Cisco SPA525G2. > > > > Thanks. > > > > Marek > > > > ------- Original Message --- > > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp jc...@sangoma.com > > wrote: > > > > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško marek.gre...@protonmail.com > > wrote: > > > > > It looks like all phones get unregistered, but I am not aware of the > > > cause. Why are get not registered when there is a connectivity between > > > them and asterisk? > > > > Are the REGISTER requests reaching Asterisk (do they show up in a packet > > capture, do they show up in "pjsip set logger on")? It needs to be further > > isolated. How are the phones configured to reach Asterisk? If using a > > hostname, are they still able to resolve it? > > > > -- > > Joshua C. Colp > > Asterisk Project Lead > > Sangoma Technologies > > Check us out at www.sangoma.com and 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 -- _ -- 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] Local calls not possible when Internet connection down
Hello Joshua, thanks for suggestion. I just found out the same solution several minutes ago. I also obtained the maintenance window, so I diasbled outgoing DNS and SIP. But I was not successful reproducing the bad state. So I ceased futher debugging attempts and set srv_lookups to no. We will see once the next massive internet outage out of my control happens, whether it helped. Thanks again Marek --- Original Message --- On Tuesday, November 7th, 2023 at 16:28, Joshua C. Colp wrote: > On Tue, Nov 7, 2023 at 11:20 AM Marek Greško > wrote: > >> Hello, >> >> well I do not ask those who only guess, but those who know what is asterisk >> expected to do when internet connectivity goes down. I did not had a chance >> to make internet not to work yet, since it is needed. But inspecting dns >> logs I found out that there started to be resolving for _sip._tcp and >> _sip._udp records for the provider's server. So apparently making hosts >> record make asterisk happy when everything works, but when there is a >> communication problem then it falls back to searching for srv records. At >> least it seems to be so for now. Moreover I found out this old thread: > > The expectation is that Asterisk continues to work. That being said there is > one case (specifically using realtime with an identify section that > references a hostname) that can cause this specific behavior where PJSIP will > block. > > Are you in that scenario? If so you CAN disable SRV records on the identify > by setting "srv_lookups" to "no". > -- > > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
Hello, well I do not ask those who only guess, but those who know what is asterisk expected to do when internet connectivity goes down. I did not had a chance to make internet not to work yet, since it is needed. But inspecting dns logs I found out that there started to be resolving for _sip._tcp and _sip._udp records for the provider's server. So apparently making hosts record make asterisk happy when everything works, but when there is a communication problem then it falls back to searching for srv records. At least it seems to be so for now. Moreover I found out this old thread: https://community.freepbx.org/t/asterisk-become-mad-when-a-dns-problem-occur/4755/10 So the problem seems to be still present. So if asterisk is not able to resolve using it's dns resolver it breaks also local communication which is complete non-sense. I am thinking of two possible workarounds: 1. If thre is a possibility to convince asterisk not to fallback to searching for srv records, it would be ideal. Is somebody aware of such options in pjsip? 2. If the first workaround is not feasible I can create rpz records for provider's A and SRV records. When I will be able to shutdown internet or at least outbound DNS, I will try to make sure my findings are correct using tcpdump. Thanks Marek Sent with Proton Mail secure email. --- Original Message --- On Tuesday, November 7th, 2023 at 0:46, Greg Troxel wrote: > Marek Greško marek.gre...@protonmail.com writes: > > > But I am not sure why this is happening. I have sip providers hostname > > in /etc/hosts file to prevent such situations. Should I reconfigure it > > not to use hosts file but rather some RPZ on DNS server? Does asterisk > > ignore hosts file? Or does it try to do some srv lookups? But in > > either case, why does this influence local calls? Local domain should > > really be resolvable. > > > You should run tcpdump on 53 and 5353 in multiple places and figure out > what it is doing, rather than asking us, who can only guess. -- _ -- 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] Local calls not possible when Internet connection down
Hello, the corresponding conf is: pbx.example.lan No Yes Yes No 3600 No No No 3600 Normal No No Marek --- Original Message --- On Tuesday, November 7th, 2023 at 0:22, Łukasz Grzywański wrote: > Could you show the phone configurations - section "Proxy and Registration" > > On Mon, 6 Nov 2023 at 23:13, Marek Greško wrote: > >> Hello, >> >> you are probably right. It should somehow be related to DNS. I just found >> out this in the storm of previous messages: >> >> WARNING[13945] taskprocessor.c: The 'dns_system_resolver_tp' task processor >> queue reached 500 scheduled tasks. >> >> But I am not sure why this is happening. I have sip providers hostname in >> /etc/hosts file to prevent such situations. Should I reconfigure it not to >> use hosts file but rather some RPZ on DNS server? Does asterisk ignore hosts >> file? Or does it try to do some srv lookups? But in either case, why does >> this influence local calls? Local domain should really be resolvable. >> >> Thanks >> >> Marek >> >> --- Original Message --- >> On Monday, November 6th, 2023 at 19:52, Marek Greško >> wrote: >> >>> Hello, >>> >>> sure I have local DNS server and public resolving should not be needed for >>> phone registrations. Running pjsip show endpojnt show the endpoints as not >>> in use. >>> >>> When looking into logs I see only res_pjsip_outbound_registration.c: No >>> response >>> received from sip provider. Nothing else. >>> >>> In phone log I see: >>> CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), >>> lid=0, par=0, par2=(nil)) >>> >>> The phone is Cisco SPA525G2. >>> >>> Thanks. >>> >>> Marek >>> >>> --- Original Message --- >>> On Monday, November 6th, 2023 at 15:45, Joshua C. Colp >>> wrote: >>> >>>> On Mon, Nov 6, 2023 at 10:42 AM Marek Greško >>>> wrote: >>>> >>>>> It looks like all phones get unregistered, but I am not aware of the >>>>> cause. Why are get not registered when there is a connectivity between >>>>> them and asterisk? >>>> >>>> Are the REGISTER requests reaching Asterisk (do they show up in a packet >>>> capture, do they show up in "pjsip set logger on")? It needs to be further >>>> isolated. How are the phones configured to reach Asterisk? If using a >>>> hostname, are they still able to resolve it? >>>> -- >>>> >>>> Joshua C. Colp >>>> Asterisk Project Lead >>>> Sangoma Technologies >>>> Check us out at www.sangoma.com and 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 > > -- > > Pozdrawiam, > > Łukasz Grzywański > Voice Architect > > Mok Yok IT Sp. z o.o. > ul. Rzeźnicza 32/33, 50-130 Wrocław, Polska > tel. +48 717227200, fax +48 717227299 > mob.: +48 517255333, e-mail: lukasz.grzywan...@mokyokit.com-- _ -- 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] Local calls not possible when Internet connection down
Hello, you are probably right. It should somehow be related to DNS. I just found out this in the storm of previous messages: WARNING[13945] taskprocessor.c: The 'dns_system_resolver_tp' task processor queue reached 500 scheduled tasks. But I am not sure why this is happening. I have sip providers hostname in /etc/hosts file to prevent such situations. Should I reconfigure it not to use hosts file but rather some RPZ on DNS server? Does asterisk ignore hosts file? Or does it try to do some srv lookups? But in either case, why does this influence local calls? Local domain should really be resolvable. Thanks Marek --- Original Message --- On Monday, November 6th, 2023 at 19:52, Marek Greško wrote: > Hello, > > sure I have local DNS server and public resolving should not be needed for > phone registrations. Running pjsip show endpojnt show the endpoints as not in > use. > > When looking into logs I see only res_pjsip_outbound_registration.c: No > response > received from sip provider. Nothing else. > > In phone log I see: > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > lid=0, par=0, par2=(nil)) > > The phone is Cisco SPA525G2. > > Thanks. > > Marek > > --- Original Message --- > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp > wrote: > >> On Mon, Nov 6, 2023 at 10:42 AM Marek Greško >> wrote: >> >>> It looks like all phones get unregistered, but I am not aware of the cause. >>> Why are get not registered when there is a connectivity between them and >>> asterisk? >> >> Are the REGISTER requests reaching Asterisk (do they show up in a packet >> capture, do they show up in "pjsip set logger on")? It needs to be further >> isolated. How are the phones configured to reach Asterisk? If using a >> hostname, are they still able to resolve it? >> -- >> >> Joshua C. Colp >> Asterisk Project Lead >> Sangoma Technologies >> Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
Hello, sure I have local DNS server and public resolving should not be needed for phone registrations. Running pjsip show endpojnt show the endpoints as not in use. When looking into logs I see only res_pjsip_outbound_registration.c: No response received from sip provider. Nothing else. In phone log I see: CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), lid=0, par=0, par2=(nil)) The phone is Cisco SPA525G2. Thanks. Marek --- Original Message --- On Monday, November 6th, 2023 at 15:45, Joshua C. Colp wrote: > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško > wrote: > >> It looks like all phones get unregistered, but I am not aware of the cause. >> Why are get not registered when there is a connectivity between them and >> asterisk? > > Are the REGISTER requests reaching Asterisk (do they show up in a packet > capture, do they show up in "pjsip set logger on")? It needs to be further > isolated. How are the phones configured to reach Asterisk? If using a > hostname, are they still able to resolve it? > -- > > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
It looks like all phones get unregistered, but I am not aware of the cause. Why are get not registered when there is a connectivity between them and asterisk? Marek --- Original Message --- On Monday, November 6th, 2023 at 15:10, Joshua C. Colp wrote: > On Mon, Nov 6, 2023 at 10:06 AM Marek Greško > wrote: > >> Hello, >> >> I just realized that when my Internet connection goes down and I loose >> connectivity to VoIP SIP provider I loose ability to make local calls after >> some time. When I restart asterisk, I am able to make local calls for some >> time, but it then suddenly stops working again. I am using pjsip stack. >> >> What could be the cause of this? > > There is insufficient information to be able to answer this. Such as, what > actually happens when attempts are made? What shows on the console? > -- > > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
Hello, I just realized that when my Internet connection goes down and I loose connectivity to VoIP SIP provider I loose ability to make local calls after some time. When I restart asterisk, I am able to make local calls for some time, but it then suddenly stops working again. I am using pjsip stack. What could be the cause of this? Thnaks Marek-- _ -- 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Hello Jerry, when you run asterisk using su, ownership of audio device files does not get updated. When you login, you get the permissions. You can verify by ls -l and getfacl on the device file. Marek --- Original Message --- On Thursday, September 14th, 2023 at 14:33, Jerry Geis wrote: > On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis wrote: > >>>An issue[1] was already created by asterisk at phreaknet.org and they also >>>put >>>a fix up for review and inclusion[2]. >> >>>[1] https://github.com/asterisk/asterisk/issues/308 >>>[2] https://github.com/asterisk/asterisk/pull/309 >> >> The change "seems" to be working. >> Will test more tomorrow - had to leave. >> THANKS! >> Jerry > > Yes - this fix is working for me. > > Only issue I have now is, I used to run asterisk like this: > su silentm -c "/usr/sbin/asterisk -fn" > I also tried > su silentm -l -c "/usr/sbin/asterisk -fn" > > these do not work for the chan_console. I have to actually login as silentm > and then run asterisks - to HEAR the audio. > doing su above I do not hear the audio - but the CLI looks the same - no > errors. > > Thoughts? > > 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] MWI with PJSIP - unsollicitated works fine, solicitated doesn't
Hello Michael, I was also struggling with solicited MWI after moving to pjsip. My problem was I was defining mailbox=111@extensioncontext. But the correct context in the mailbox command is to be defined by context in voicemail.conf. My voicemails were all defined in the context default (see voicemail.conf) and the mailbox command should look like this: mailbox=111@default. Hope this helps. I do not know whether this is also your problem. Marek 2021-11-14 16:38 GMT+01:00, Mike : > Hi, > > > > Just recently moved over from chan_sip to PJSIP and am slowly cleaning up > whatever needs to be. > > > > I can't seem to make sollicitated MWI work, but unsollicitated works fine. > > > > > I got my phones subscribing to mailbox@context (i.e. 100@whatever) > > > > I have my related AOR entry (realtime, in a DB) set to > mailboxes=100@whatever . I can see it is set properly by using the command > "pjsip show aor " > > > > But when I turn pjsip logger on, I see messages from the phones > subscribing and SIP/2.0 401 Unauthorized messages back. > > > > If I put the same column in my realtime DB (mailboxes) for ENPOINT to the > same value (100@whatever) then it works fine, MWI works on the phone. > > > > For a few reasons I'd like to get MWI working in sollicitated mode > instead. Is there a trick to it? > > > > I upgraded to Asterisk 18.8.0 just to see if a later patch fixed anything, > so I am current. > > > > > > > > > > > > > > Michael > > -- _ -- 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] Couldn't find auth 'provider'
Hello, I rewritten the auth section once more manually and disabled the stir shaken modules and now it works. I do not expect stir shaken modules could cause such issues so there should have been some unseen white characters in the configuration of auth section or something like that. Strange the section worked for registration. Thanks for you effort. Marek ut 19. 10. 2021 o 20:22 Joshua C. Colp napísal(a): > On Tue, Oct 19, 2021 at 3:18 PM Marek Greško wrote: > >> Hello, >> >> I am observing error: >> res_pjsip/pjsip_configuration.c:2368 ast_sip_retrieve_auths: Couldn't >> find auth 'provider'. Cannot authenticate. >> res_pjsip_outbound_authenticator_digest.c:144 >> digest_create_request_with_auth: Endpoint: 'provider': Failed to set >> authentication credentials >> >> I use config below. It reports the auth section is missing, but it is >> apparently here. >> >> What am I doing wrong? >> > > What does "pjsip show auths" in the CLI show? When PJSIP loads does it > state an error with the configuration or any configuration? What is the > contents of sorcery.conf (is it trying to pull auths from elsewhere)? > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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
Re: [asterisk-users] Couldn't find auth 'provider'
Hello, pjsip show auths shows only phone accounts. nothing about provider. But strange registration works and it uses the same auth section. The sorcery.conf file contains: [test_sorcery_section] test=memory [test_sorcery_cache] test/cache=test test=memory I can see no relevant errors in /var/log/asterisk/messages file. Only some unrelated stuff: [Oct 19 19:55:18] Asterisk 18.2.0 built by mockbuild @ buildhw-x86-04.iad2.fedoraproject.org on a x86_64 running Linux on 2021-02-08 08:28:42 UTC [Oct 19 19:55:18] NOTICE[4385] loader.c: 300 modules will be loaded. [Oct 19 19:55:18] NOTICE[4385] cdr.c: CDR simple logging enabled. [Oct 19 19:55:18] WARNING[4385] res_musiconhold.c: No music on hold classes configured, disabling music on hold. [Oct 19 19:55:18] WARNING[4385] res_phoneprov.c: Unable to find a valid server address or name. [Oct 19 19:55:20] NOTICE[4385] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Oct 19 19:55:20] ERROR[4385] ari/config.c: No configured users for ARI [Oct 19 19:55:20] NOTICE[4385] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge [Oct 19 19:55:20] NOTICE[4385] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. [Oct 19 19:55:20] WARNING[4385] loader.c: Some non-required modules failed to load. [Oct 19 19:55:20] ERROR[4385] loader.c: Failed to resolve dependencies for res_stir_shaken [Oct 19 19:55:20] ERROR[4385] loader.c: res_stir_shaken declined to load. [Oct 19 19:55:20] ERROR[4385] loader.c: Failed to resolve dependencies for res_pjsip_stir_shaken [Oct 19 19:55:20] ERROR[4385] loader.c: res_pjsip_stir_shaken declined to load. Thanks Marek ut 19. 10. 2021 o 20:22 Joshua C. Colp napísal(a): > On Tue, Oct 19, 2021 at 3:18 PM Marek Greško wrote: > >> Hello, >> >> I am observing error: >> res_pjsip/pjsip_configuration.c:2368 ast_sip_retrieve_auths: Couldn't >> find auth 'provider'. Cannot authenticate. >> res_pjsip_outbound_authenticator_digest.c:144 >> digest_create_request_with_auth: Endpoint: 'provider': Failed to set >> authentication credentials >> >> I use config below. It reports the auth section is missing, but it is >> apparently here. >> >> What am I doing wrong? >> > > What does "pjsip show auths" in the CLI show? When PJSIP loads does it > state an error with the configuration or any configuration? What is the > contents of sorcery.conf (is it trying to pull auths from elsewhere)? > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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] Couldn't find auth 'provider'
Hello, I am observing error: res_pjsip/pjsip_configuration.c:2368 ast_sip_retrieve_auths: Couldn't find auth 'provider'. Cannot authenticate. res_pjsip_outbound_authenticator_digest.c:144 digest_create_request_with_auth: Endpoint: 'provider': Failed to set authentication credentials I use config below. It reports the auth section is missing, but it is apparently here. What am I doing wrong? Thanks Marek ; [global] type = global debug = no [transport-udp] type = transport protocol = udp bind = 0.0.0.0 external_media_address = x.x.x.x external_signaling_address = x.x.x.x local_net =192.168.1.0/255.255.255.0 [provider] type = registration transport = transport-udp outbound_auth = provider server_uri = sip:... client_uri = sip:... contact_user = username retry_interval = 20 forbidden_retry_interval = 600 expiration = 120 ;max_retries = 10 [provider] type = auth auth_type = userpass username = username password = password [provider] type = endpoint context = provider-in dtmf_mode = rfc4733 ;direct_media = no from_domain = provider.domain force_rport = yes rewrite_contact = yes rtp_symmetric = yes allow_subscribe = no outbound_auth = provider aors = provider disallow = all allow = alaw allow = ilbc allow = g729 allow = gsm allow = g723 [provider] type = aor contact = sip:... qualify_frequency = 15 [provider] type = identify endpoint = provider match = ... -- _ -- 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, I already read the scenario you pointed to. It is not really the same. as you can see in my rules I sent before I have CT in both directions. Related to configuration error I am 99% sure the configuration is correct. It was generated by automatic tool and then slightly edited and reviewed by nftables guru. I just admit the there could be some configuration error. Maybe some race condition in systemd - wrong dependencies or something like that. I do not know. But I am sure once I will find it (or suffer longer). The reason many people use it and they will notice is invalid. I hit a bug in PMTU dicovery several moths ago. And no one was complaining at all. The bug is now fixed, so it is pretty probable it is a bug. The reasoning that no expliot has been found in rtp for 20 year is invalid. We are not talking about bugs in rtp. We are talking about open ports and application local to asterisk server could use. So many backdoors can be open. Believe me. It is not secure. Maybe it is acceptable on a dedicated asterisk box, but not on a multi purpose server. Marek 2021-09-10 23:28 GMT+02:00, Duncan Turnbull : > > >> On 11/09/2021, at 2:54 AM, Marek Greško wrote: >> >> Hello, >> >> thanks you very much for your effort. Without your help I would never >> realize the problem lies in the firewall. >> >> But what do you mean by the doubt that it is bug? You mean it should >> be configured another way? I do not claim my configuration is correct. >> I am also new to nftables. But I do not think opening the wide port >> range is a solution. The nftables runs on the asterisk server itself. > > The reason I don’t use sip algs is because they have a have a function that > isn’t required. And a complexity that messes things up. No exploit has yet > been found for rtp for 20 years and it has been open to the world. For > whatever reason you can’t get your head around this being a valid option so > then you are jumping to a bug when you freely admit your lack of familiarity > > > This may be your scenario > > https://unix.stackexchange.com/questions/461320/nf-conntrack-sip-does-not-work-sometimes-restarting-iptables-usually-fixes-it > > You are adding a dependency on the firewall that you don’t need using > configuration you are not sure of. That is never a reliable situation to be > in. > > Why would nftables have a bug? Many people use it around the world and it > works well. What is the likelihood of a bug in this scenario > > The alternative is a misconfiguration, and you are not very familiar with > the configuration and new to nftables. Which one is more likely? > > The above issue sounds like yours but it could be something else > > You can research and find the config error, or somehow you can prove a bug > or you can remove the issue by just allowing rtp through > > All of these are your choices. To me the config error is most likely as I > have very rarely found a bug. It’s almost always config > >> >> Marek >> >> >> 2021-09-10 1:19 GMT+02:00, Duncan Turnbull : >>> >>> >>>>> On 10/09/2021, at 4:37 AM, Marek Greško wrote: >>>> >>>> There are other systems running on the same hardware. It would just >>>> leave open ports here. >>>> >>>> Do not compare SIP ALG on a closed source device to an opensource >>>> software with active development. I had no such problems in the past >>>> when using iptables. The nftables is a pretty new software, so some >>>> bugs could be present and I accept. I just wanted to be sure I am not >>>> doing anything wrong. Now I am pretty sure it is a bug. >>> >>> I very much doubt it’s a bug, but that’s your choice to pursue that >>> >>> You ask for help but perhaps you are not wanting to listen >>> >>> If you open your asterisk rtp ports in your firewall then you are >>> following >>> pretty much what everyone else does. >>> >>> Otherwise you are letting another device interfere with your Sip >>> transactions and we have already shown that’s a bad idea. Makes no >>> difference whether it’s open source or not. >>> >>> But up to you >>> >>>> >>>> Thanks >>>> >>>> Marek >>>> >>>> >>>> 2021-09-09 18:30 GMT+02:00, Administrator : >>>>> >>>>>> Le 09/09/2021 à 18:15, Marek Greško a écrit : >>>>>> There is always some risk. If there is a solution that should work, >>>>>> it >>>>>> is best to use it. We just need the root cause, why it fail
Re: [asterisk-users] problems with natted phones
Hello, thanks you very much for your effort. Without your help I would never realize the problem lies in the firewall. But what do you mean by the doubt that it is bug? You mean it should be configured another way? I do not claim my configuration is correct. I am also new to nftables. But I do not think opening the wide port range is a solution. The nftables runs on the asterisk server itself. Marek 2021-09-10 1:19 GMT+02:00, Duncan Turnbull : > > >> On 10/09/2021, at 4:37 AM, Marek Greško wrote: >> >> There are other systems running on the same hardware. It would just >> leave open ports here. >> >> Do not compare SIP ALG on a closed source device to an opensource >> software with active development. I had no such problems in the past >> when using iptables. The nftables is a pretty new software, so some >> bugs could be present and I accept. I just wanted to be sure I am not >> doing anything wrong. Now I am pretty sure it is a bug. > > I very much doubt it’s a bug, but that’s your choice to pursue that > > You ask for help but perhaps you are not wanting to listen > > If you open your asterisk rtp ports in your firewall then you are following > pretty much what everyone else does. > > Otherwise you are letting another device interfere with your Sip > transactions and we have already shown that’s a bad idea. Makes no > difference whether it’s open source or not. > > But up to you > >> >> Thanks >> >> Marek >> >> >> 2021-09-09 18:30 GMT+02:00, Administrator : >>> >>>> Le 09/09/2021 à 18:15, Marek Greško a écrit : >>>> There is always some risk. If there is a solution that should work, it >>>> is best to use it. We just need the root cause, why it fails >>>> sometimes. >>> >>> Like SIP ALG ? ;) Please explain which risk are existing if there is >>> nothing listening on those ports ? >>> >>>> >>>> >>>> 2021-09-09 18:01 GMT+02:00, Antony Stone >>>> : >>>>> On Thursday 09 September 2021 at 17:56:10, Marek Greško wrote: >>>>> >>>>>> Hello, >>>>>> >>>>>> I would not like to open whole range of udp ports for rtp. >>>>> Why not? What is the risk? >>>>> >>>>> What would possibly be listening on UDP ports 1 - 2 (the >>>>> Asterisk >>>>> default range) which an external scanner / attacker could make use of? >>> >>> -- >>> Daniel >>> >>> -- >>> _ >>> -- 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 -- _ -- 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
There are other systems running on the same hardware. It would just leave open ports here. Do not compare SIP ALG on a closed source device to an opensource software with active development. I had no such problems in the past when using iptables. The nftables is a pretty new software, so some bugs could be present and I accept. I just wanted to be sure I am not doing anything wrong. Now I am pretty sure it is a bug. Thanks Marek 2021-09-09 18:30 GMT+02:00, Administrator : > > Le 09/09/2021 à 18:15, Marek Greško a écrit : >> There is always some risk. If there is a solution that should work, it >> is best to use it. We just need the root cause, why it fails >> sometimes. > > Like SIP ALG ? ;) Please explain which risk are existing if there is > nothing listening on those ports ? > >> >> >> 2021-09-09 18:01 GMT+02:00, Antony Stone >> : >>> On Thursday 09 September 2021 at 17:56:10, Marek Greško wrote: >>> >>>> Hello, >>>> >>>> I would not like to open whole range of udp ports for rtp. >>> Why not? What is the risk? >>> >>> What would possibly be listening on UDP ports 1 - 2 (the Asterisk >>> default range) which an external scanner / attacker could make use of? > > -- > Daniel > > -- > _ > -- 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] problems with natted phones
There is always some risk. If there is a solution that should work, it is best to use it. We just need the root cause, why it fails sometimes. Marek 2021-09-09 18:01 GMT+02:00, Antony Stone : > On Thursday 09 September 2021 at 17:56:10, Marek Greško wrote: > >> Hello, >> >> I would not like to open whole range of udp ports for rtp. > > Why not? What is the risk? > > What would possibly be listening on UDP ports 1 - 2 (the Asterisk > default range) which an external scanner / attacker could make use of? > > > Antony. > > -- > Too many people spend money they haven't earned > to buy things they don't want, > to impress people they don't like. > > - Will Rogers > >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
Re: [asterisk-users] problems with natted phones
Hello,
I would not like to open whole range of udp ports for rtp. I use
nf_conntrack_sip module for dynamically opening relevant ports. And
there is probably some bug in it.
Marek
2021-09-08 23:12 GMT+02:00, Administrator :
> Hi. Our rules:
>
> Le 08/09/2021 à 22:43, Marek Greško a écrit :
>> Hello,
>>
>> I did converted from iptables by automatical script and then rewritten
>> myself, because not everything was rewritten successfully.
>>
>> Relevant parts:
>>
>> table ip filter {
>>ct helper sip {
>> type "sip" protocol udp
>> l3proto ip
>>}
>>
>>chain PREROUTING {
>> type filter hook prerouting priority filter; policy accept;
>> udp port 5060 ct helper set "sip"
>>}
>>
>>chain INPUT {
>> ...
>> ct state invalid drop
>> ct state related accept
>> ct state established accept
>> ...
>> iifname "ppp0" jump i-inet
>>}
>
> set world_udp.eth0 {
> type inet_service
> flags interval
> elements = { iax, sip, sip-tls, 1-3 }
> }
>
> chain input {
> type filter hook input priority 0; policy drop;
> iif "eth0" ip daddr udp dport
> @world_udp.eth0 counter packets 15394440 bytes 3738156190 accept
>
>
> As you see we take care on RTP port range defined in rtp,conf
>
>>
>>chain OUTPUT {
>> type filter hook output priority filter; policy accept;
>> udp port 5060 ct helper set "sip"
>> ...
>>}
> chain output {
> type filter hook output priority 0; policy drop;
> oif "eth0" ct state established,related,new counter
> packets 17542533 bytes 6033494909 accept
>
> our default policy is to drop so we add new in ct state
>
>>
>>chain i-inet {
>> ...
>> udp port 5060 jump r-sip
>> ...
>>}
>>
>>chain r-sip {
>> ip saddr 192.0.2.0/24 accept
>>}
>> }
>>
>> table ip mangle {
>>chain PREROUTING {
>> type filter hook prerouting priority mangle; policy accept;
>> ...
>> udp sport 5060 ip dscp set 0x04
>>}
>>
>>chain OUTPUT {
>> type route hook output priority mangle; policy accept;
>> ...
>> udp dport 5060 ip dscp set 0x04
>> ...
>>}
>> }
>>
>> table ip6 filter {
>>ct helper sip {
>> type "sip" protocol udp
>> l3proto ip6
>>}
>>
>>... pretty the same, but I have no ipv6 internet connectivity, so
>> this should not match ...
>>
>> }
>>
>>
>> Is there something incorrect?
>>
>> Thanks
>>
>> Marek
>>
>>
>>
>> 2021-09-08 21:17 GMT+02:00, Duncan Turnbull :
>>>
>>>> On 9/09/2021, at 6:23 AM, Marek Greško wrote:
>>>>
>>>> Hello,
>>>>
>>>> I confirm temporarily allowing all the udp communication from the nat
>>>> ip address solved the problem, so the problem lies in the nftables.
>>>> This is probably not the right forum to continue. Or is it? Does
>>>> anybody have wide experience with nftables and sip?
>>> If you publish your rule set then we could look. Did you write the rules?
>>> What have you checked so far?
>>>
>>>> Thanks
>>>>
>>>> Marek
>>>>
>>>>
>>>> 2021-09-07 10:40 GMT+02:00, Antony Stone
>>>> :
>>>>> On Monday 06 September 2021 at 23:05:27, Duncan Turnbull wrote:
>>>>>
>>>>>>>> On 7/09/2021, at 8:30 AM, Marek Greško wrote:
>>>>>>>>
>>>>>>>> Hello,
>>>>>>>>
>>>>>>>> it is only local nftables with nf_conntrack_sip on the asterisk
>>>>>>>> server. Probably a kernel bug? It did not trigger with previous
>>>>>>>> providers since they had working SIP ALG. Now I hear no audio in
>>>>>>>> both
>>>>>>>> directions because outgoing rtp stream from asterisk goes to private
>>>>>>>> address space and incoming stream is blocked. So the outgoing rtp
>>>>>>>> could not be learnt to send to nat addess.
>>>>>> Maybe a bug but that’s less li
Re: [asterisk-users] problems with natted phones
Sorry did convert, not did converted :)
2021-09-08 22:43 GMT+02:00, Marek Greško :
> Hello,
>
> I did converted from iptables by automatical script and then rewritten
> myself, because not everything was rewritten successfully.
>
> Relevant parts:
>
> table ip filter {
> ct helper sip {
> type "sip" protocol udp
> l3proto ip
> }
>
> chain PREROUTING {
> type filter hook prerouting priority filter; policy accept;
> udp port 5060 ct helper set "sip"
> }
>
> chain INPUT {
> ...
> ct state invalid drop
> ct state related accept
> ct state established accept
> ...
> iifname "ppp0" jump i-inet
> }
>
> chain OUTPUT {
> type filter hook output priority filter; policy accept;
> udp port 5060 ct helper set "sip"
> ...
> }
>
> chain i-inet {
> ...
> udp port 5060 jump r-sip
> ...
> }
>
> chain r-sip {
> ip saddr 192.0.2.0/24 accept
> }
> }
>
> table ip mangle {
> chain PREROUTING {
> type filter hook prerouting priority mangle; policy accept;
> ...
> udp sport 5060 ip dscp set 0x04
> }
>
> chain OUTPUT {
> type route hook output priority mangle; policy accept;
> ...
> udp dport 5060 ip dscp set 0x04
> ...
> }
> }
>
> table ip6 filter {
> ct helper sip {
> type "sip" protocol udp
> l3proto ip6
> }
>
> ... pretty the same, but I have no ipv6 internet connectivity, so
> this should not match ...
>
> }
>
>
> Is there something incorrect?
>
> Thanks
>
> Marek
>
>
>
> 2021-09-08 21:17 GMT+02:00, Duncan Turnbull :
>>
>>
>>> On 9/09/2021, at 6:23 AM, Marek Greško wrote:
>>>
>>> Hello,
>>>
>>> I confirm temporarily allowing all the udp communication from the nat
>>> ip address solved the problem, so the problem lies in the nftables.
>>> This is probably not the right forum to continue. Or is it? Does
>>> anybody have wide experience with nftables and sip?
>> If you publish your rule set then we could look. Did you write the rules?
>> What have you checked so far?
>>
>>>
>>> Thanks
>>>
>>> Marek
>>>
>>>
>>> 2021-09-07 10:40 GMT+02:00, Antony Stone
>>> :
>>>> On Monday 06 September 2021 at 23:05:27, Duncan Turnbull wrote:
>>>>
>>>>>>> On 7/09/2021, at 8:30 AM, Marek Greško wrote:
>>>>>>>
>>>>>>> Hello,
>>>>>>>
>>>>>>> it is only local nftables with nf_conntrack_sip on the asterisk
>>>>>>> server. Probably a kernel bug? It did not trigger with previous
>>>>>>> providers since they had working SIP ALG. Now I hear no audio in
>>>>>>> both
>>>>>>> directions because outgoing rtp stream from asterisk goes to private
>>>>>>> address space and incoming stream is blocked. So the outgoing rtp
>>>>>>> could not be learnt to send to nat addess.
>>>>>
>>>>> Maybe a bug but that’s less likely than a config error. Time to debug
>>>>> your
>>>>> nftables.
>>>>
>>>> Try temporarily simply turning the firewall off - allow all traffic
>>>> through
>>>> (although leave in place any NAT rules).
>>>>
>>>> If you then find that RTP works, you know where the problem lies.
>>>>
>>>>
>>>> Antony.
>>>>
>>>> --
>>>> Perfection in design is achieved not when there is nothing left to add,
>>>> but
>>>> rather when there is nothing left to take away.
>>>>
>>>> - Antoine de Saint-Exupery
>>>>
>>>> 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.o
Re: [asterisk-users] problems with natted phones
Hello,
I did converted from iptables by automatical script and then rewritten
myself, because not everything was rewritten successfully.
Relevant parts:
table ip filter {
ct helper sip {
type "sip" protocol udp
l3proto ip
}
chain PREROUTING {
type filter hook prerouting priority filter; policy accept;
udp port 5060 ct helper set "sip"
}
chain INPUT {
...
ct state invalid drop
ct state related accept
ct state established accept
...
iifname "ppp0" jump i-inet
}
chain OUTPUT {
type filter hook output priority filter; policy accept;
udp port 5060 ct helper set "sip"
...
}
chain i-inet {
...
udp port 5060 jump r-sip
...
}
chain r-sip {
ip saddr 192.0.2.0/24 accept
}
}
table ip mangle {
chain PREROUTING {
type filter hook prerouting priority mangle; policy accept;
...
udp sport 5060 ip dscp set 0x04
}
chain OUTPUT {
type route hook output priority mangle; policy accept;
...
udp dport 5060 ip dscp set 0x04
...
}
}
table ip6 filter {
ct helper sip {
type "sip" protocol udp
l3proto ip6
}
... pretty the same, but I have no ipv6 internet connectivity, so
this should not match ...
}
Is there something incorrect?
Thanks
Marek
2021-09-08 21:17 GMT+02:00, Duncan Turnbull :
>
>
>> On 9/09/2021, at 6:23 AM, Marek Greško wrote:
>>
>> Hello,
>>
>> I confirm temporarily allowing all the udp communication from the nat
>> ip address solved the problem, so the problem lies in the nftables.
>> This is probably not the right forum to continue. Or is it? Does
>> anybody have wide experience with nftables and sip?
> If you publish your rule set then we could look. Did you write the rules?
> What have you checked so far?
>
>>
>> Thanks
>>
>> Marek
>>
>>
>> 2021-09-07 10:40 GMT+02:00, Antony Stone
>> :
>>> On Monday 06 September 2021 at 23:05:27, Duncan Turnbull wrote:
>>>
>>>>>> On 7/09/2021, at 8:30 AM, Marek Greško wrote:
>>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> it is only local nftables with nf_conntrack_sip on the asterisk
>>>>>> server. Probably a kernel bug? It did not trigger with previous
>>>>>> providers since they had working SIP ALG. Now I hear no audio in both
>>>>>> directions because outgoing rtp stream from asterisk goes to private
>>>>>> address space and incoming stream is blocked. So the outgoing rtp
>>>>>> could not be learnt to send to nat addess.
>>>>
>>>> Maybe a bug but that’s less likely than a config error. Time to debug
>>>> your
>>>> nftables.
>>>
>>> Try temporarily simply turning the firewall off - allow all traffic
>>> through
>>> (although leave in place any NAT rules).
>>>
>>> If you then find that RTP works, you know where the problem lies.
>>>
>>>
>>> Antony.
>>>
>>> --
>>> Perfection in design is achieved not when there is nothing left to add,
>>> but
>>> rather when there is nothing left to take away.
>>>
>>> - Antoine de Saint-Exupery
>>>
>>> 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
>
> --
> __
Re: [asterisk-users] problems with natted phones
Hello, I confirm temporarily allowing all the udp communication from the nat ip address solved the problem, so the problem lies in the nftables. This is probably not the right forum to continue. Or is it? Does anybody have wide experience with nftables and sip? Thanks Marek 2021-09-07 10:40 GMT+02:00, Antony Stone : > On Monday 06 September 2021 at 23:05:27, Duncan Turnbull wrote: > >> > On 7/09/2021, at 8:30 AM, Marek Greško wrote: >> > >> > Hello, >> > >> > it is only local nftables with nf_conntrack_sip on the asterisk >> > server. Probably a kernel bug? It did not trigger with previous >> > providers since they had working SIP ALG. Now I hear no audio in both >> > directions because outgoing rtp stream from asterisk goes to private >> > address space and incoming stream is blocked. So the outgoing rtp >> > could not be learnt to send to nat addess. >> >> Maybe a bug but that’s less likely than a config error. Time to debug your >> nftables. > > Try temporarily simply turning the firewall off - allow all traffic through > (although leave in place any NAT rules). > > If you then find that RTP works, you know where the problem lies. > > > Antony. > > -- > Perfection in design is achieved not when there is nothing left to add, but > rather when there is nothing left to take away. > > - Antoine de Saint-Exupery > >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
Re: [asterisk-users] problems with natted phones
Hello, it is only local nftables with nf_conntrack_sip on the asterisk server. Probably a kernel bug? It did not trigger with previous providers since they had working SIP ALG. Now I hear no audio in both directions because outgoing rtp stream from asterisk goes to private address space and incoming stream is blocked. So the outgoing rtp could not be learnt to send to nat addess. Marek 2021-09-06 22:17 GMT+02:00, Duncan Turnbull : > > >> On 7/09/2021, at 3:08 AM, Marek Greško wrote: >> >> Hello, >> >> so when debugging RTP in asterisk there was no rtp income from the >> remote site. I did check remote nat ip address and it was same as the >> one in the pjsip show aors. So it is not due to ip address change. It >> seems the local firewall sip module does not allow rtp stream to get >> into. It was working previously with the other provider because of >> working SIP ALG on their gateways. But now with this provider and >> disabled SIP ALG it is not allowed. As I remeber in the past these >> setups did work. What are your experiences on this? >> > You would need to provide a lot more explanation here. What is your > firewall? I am assuming you configure it so find the configuration that’s > blocking the ports and change it. > > My experience as before was that something is blocking rtp, now you know > what that something is and it’s under your control so you need to check it’s > configuration and fix it. I don’t use a sip firewall. If I have external sip > clients I use a proxy. > >> Thanks >> >> Marek >> >> >> 2021-09-06 11:50 GMT+02:00, Marek Greško : >>> Sorry rtp set debug on showed something. So let try for the problem to >>> arise again. >>> >>> Marek >>> >>> >>> 2021-09-06 11:48 GMT+02:00, Marek Greško : >>>> Hello, >>>> >>>>>> I would expect that when asterisk is aware of nat, it does not send >>>>>> the rtp until it receives rtp from other side to learn the port, but >>>>>> OK, no problem to accept the behavior. >>>>>> >>>>> That’s not how things work. You should google how sip rtp and Nat work >>>>> as >>>>> it >>>>> will help you >>>> >>>> no problem if it is intended. >>>> >>>>>> >>>>>>> The question is why your asterisk didn't learn the external address >>>>>>> and >>>>>>> port from the received rtp packet >>>>>>> >>>>>>> You can look at your logs with debug to see what decisions its >>>>>>> making. >>>>>>> You >>>>>>> can see if different rtp ports have different results. >>>>>>> Your phone provider has rtp on 5010 unsuccessfully and 5016 >>>>>>> successfully. >>>>>>> Your asterisk uses rtp 13786 successfully and fails when using 18892. >>>>>>> Is >>>>>>> it >>>>>>> possible your firewall is blocking port 18892 and so asterisk never >>>>>>> sees >>>>>>> the returned packet and can't learn from it? >>>>>> >>>>>> It is very unprobable. I see no reason for blocking the port. The >>>>>> problem is asterisk never learns the correct port, so there is nothing >>>>>> to block. >>>>> It wasn’t what is probable, look at the asterisk logs and see what it’s >>>>> actually doing. If asterisk never sees the reply then you will know >>>>> something is blocking or stealing the port for some other service >>>> >>>> If it is stolen port for rtp, the next call would solve it, since it >>>> will use different one, and it does not solve it. >>>> >>>>>> >>>>>>> >>>>>>> In any event you should put your debug on and look at your logs in >>>>>>> asterisk >>>>>>> to see what it sees and why it doesn't react to the rtp packet, if it >>>>>>> gets >>>>>>> it >>>>>> >>>>>> Could you point me how the debug should be conducted? >>>>> >>>>> Using the asterisk cli turn on debug for the peer and rtp and see what >>>>> happens. Match it with the asterisk processes. You have to do this, you >>>>> can >>>>> look at cli or the log files, follow it through to see
Re: [asterisk-users] problems with natted phones
Hello, so when debugging RTP in asterisk there was no rtp income from the remote site. I did check remote nat ip address and it was same as the one in the pjsip show aors. So it is not due to ip address change. It seems the local firewall sip module does not allow rtp stream to get into. It was working previously with the other provider because of working SIP ALG on their gateways. But now with this provider and disabled SIP ALG it is not allowed. As I remeber in the past these setups did work. What are your experiences on this? Thanks Marek 2021-09-06 11:50 GMT+02:00, Marek Greško : > Sorry rtp set debug on showed something. So let try for the problem to > arise again. > > Marek > > > 2021-09-06 11:48 GMT+02:00, Marek Greško : >> Hello, >> >>>> I would expect that when asterisk is aware of nat, it does not send >>>> the rtp until it receives rtp from other side to learn the port, but >>>> OK, no problem to accept the behavior. >>>> >>> That’s not how things work. You should google how sip rtp and Nat work >>> as >>> it >>> will help you >> >> no problem if it is intended. >> >>>> >>>>> The question is why your asterisk didn't learn the external address >>>>> and >>>>> port from the received rtp packet >>>>> >>>>> You can look at your logs with debug to see what decisions its making. >>>>> You >>>>> can see if different rtp ports have different results. >>>>> Your phone provider has rtp on 5010 unsuccessfully and 5016 >>>>> successfully. >>>>> Your asterisk uses rtp 13786 successfully and fails when using 18892. >>>>> Is >>>>> it >>>>> possible your firewall is blocking port 18892 and so asterisk never >>>>> sees >>>>> the returned packet and can't learn from it? >>>> >>>> It is very unprobable. I see no reason for blocking the port. The >>>> problem is asterisk never learns the correct port, so there is nothing >>>> to block. >>> It wasn’t what is probable, look at the asterisk logs and see what it’s >>> actually doing. If asterisk never sees the reply then you will know >>> something is blocking or stealing the port for some other service >> >> If it is stolen port for rtp, the next call would solve it, since it >> will use different one, and it does not solve it. >> >>>> >>>>> >>>>> In any event you should put your debug on and look at your logs in >>>>> asterisk >>>>> to see what it sees and why it doesn't react to the rtp packet, if it >>>>> gets >>>>> it >>>> >>>> Could you point me how the debug should be conducted? >>> >>> Using the asterisk cli turn on debug for the peer and rtp and see what >>> happens. Match it with the asterisk processes. You have to do this, you >>> can >>> look at cli or the log files, follow it through to see the rtp packet >>> being >>> received. Lots of debug advice on google. >> >> Asterisk cli did not show anything interesting. I tried pjsip set >> logger verbose on, but no logs showed anywhere. What am I doing wrong? >> >> Marek >> >> >>>> >>>> Is my suspection that the problem could be caused by nat ip addres >>>> changing reasonable? How should asterisk handle the situation? >>> I can’t see anything to support that. Everything is looking normal >>> except >>> asterisk doesn’t appear to beseeing the rtp packet >>>> >>>> Thanks >>>> >>>> Marek >>>> >>>> >>>>> >>>>> Have fun, its all good learning. >>>>> >>>>> >>>>>> On Sun, Sep 5, 2021 at 6:27 PM Marek Greško >>>>>> wrote: >>>>>> >>>>>> Hello, >>>>>> >>>>>> regarding the ipv6, you see nothing about that it should be some type >>>>>> of ipv6 tunnelling, because also MTU is lower than expected. You >>>>>> should not see any ipv6 related communication in the sniff. Phone is >>>>>> not aware of it. >>>>>> >>>>>> The asterisk's static public ip address is 198.51.100.1. >>>>>> The remote provider's dynamic nat pool is 192.0.2.0/24. By provider >>>>>> we >>>>>> mean internet provider the rem
Re: [asterisk-users] problems with natted phones
Sorry rtp set debug on showed something. So let try for the problem to arise again. Marek 2021-09-06 11:48 GMT+02:00, Marek Greško : > Hello, > >>> I would expect that when asterisk is aware of nat, it does not send >>> the rtp until it receives rtp from other side to learn the port, but >>> OK, no problem to accept the behavior. >>> >> That’s not how things work. You should google how sip rtp and Nat work as >> it >> will help you > > no problem if it is intended. > >>> >>>> The question is why your asterisk didn't learn the external address and >>>> port from the received rtp packet >>>> >>>> You can look at your logs with debug to see what decisions its making. >>>> You >>>> can see if different rtp ports have different results. >>>> Your phone provider has rtp on 5010 unsuccessfully and 5016 >>>> successfully. >>>> Your asterisk uses rtp 13786 successfully and fails when using 18892. >>>> Is >>>> it >>>> possible your firewall is blocking port 18892 and so asterisk never >>>> sees >>>> the returned packet and can't learn from it? >>> >>> It is very unprobable. I see no reason for blocking the port. The >>> problem is asterisk never learns the correct port, so there is nothing >>> to block. >> It wasn’t what is probable, look at the asterisk logs and see what it’s >> actually doing. If asterisk never sees the reply then you will know >> something is blocking or stealing the port for some other service > > If it is stolen port for rtp, the next call would solve it, since it > will use different one, and it does not solve it. > >>> >>>> >>>> In any event you should put your debug on and look at your logs in >>>> asterisk >>>> to see what it sees and why it doesn't react to the rtp packet, if it >>>> gets >>>> it >>> >>> Could you point me how the debug should be conducted? >> >> Using the asterisk cli turn on debug for the peer and rtp and see what >> happens. Match it with the asterisk processes. You have to do this, you >> can >> look at cli or the log files, follow it through to see the rtp packet >> being >> received. Lots of debug advice on google. > > Asterisk cli did not show anything interesting. I tried pjsip set > logger verbose on, but no logs showed anywhere. What am I doing wrong? > > Marek > > >>> >>> Is my suspection that the problem could be caused by nat ip addres >>> changing reasonable? How should asterisk handle the situation? >> I can’t see anything to support that. Everything is looking normal except >> asterisk doesn’t appear to beseeing the rtp packet >>> >>> Thanks >>> >>> Marek >>> >>> >>>> >>>> Have fun, its all good learning. >>>> >>>> >>>>> On Sun, Sep 5, 2021 at 6:27 PM Marek Greško >>>>> wrote: >>>>> >>>>> Hello, >>>>> >>>>> regarding the ipv6, you see nothing about that it should be some type >>>>> of ipv6 tunnelling, because also MTU is lower than expected. You >>>>> should not see any ipv6 related communication in the sniff. Phone is >>>>> not aware of it. >>>>> >>>>> The asterisk's static public ip address is 198.51.100.1. >>>>> The remote provider's dynamic nat pool is 192.0.2.0/24. By provider we >>>>> mean internet provider the remote phones are behind. We are not >>>>> complaining about voip provider, we have no problem with that. Only >>>>> communication between asterisk and remote phones behind some internet >>>>> provider. This is the only conversation to look at. >>>>> The phone private address is 192.168.100.235. >>>>> >>>>> Thanks >>>>> >>>>> Marek >>>>> >>>>> >>>>> 2021-09-05 1:11 GMT+02:00, Duncan Turnbull : >>>>>> >>>>>> >>>>>>> On 5/09/2021, at 10:21 AM, Marek Greško wrote: >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> could you please answer my previous question about anonymizing >>>>>>> several >>>>>>> parameters? I have the data ready, but will post after answer. I >>>>>>> have >>>>&g
Re: [asterisk-users] problems with natted phones
Hello, >> I would expect that when asterisk is aware of nat, it does not send >> the rtp until it receives rtp from other side to learn the port, but >> OK, no problem to accept the behavior. >> > That’s not how things work. You should google how sip rtp and Nat work as it > will help you no problem if it is intended. >> >>> The question is why your asterisk didn't learn the external address and >>> port from the received rtp packet >>> >>> You can look at your logs with debug to see what decisions its making. >>> You >>> can see if different rtp ports have different results. >>> Your phone provider has rtp on 5010 unsuccessfully and 5016 successfully. >>> Your asterisk uses rtp 13786 successfully and fails when using 18892. Is >>> it >>> possible your firewall is blocking port 18892 and so asterisk never sees >>> the returned packet and can't learn from it? >> >> It is very unprobable. I see no reason for blocking the port. The >> problem is asterisk never learns the correct port, so there is nothing >> to block. > It wasn’t what is probable, look at the asterisk logs and see what it’s > actually doing. If asterisk never sees the reply then you will know > something is blocking or stealing the port for some other service If it is stolen port for rtp, the next call would solve it, since it will use different one, and it does not solve it. >> >>> >>> In any event you should put your debug on and look at your logs in >>> asterisk >>> to see what it sees and why it doesn't react to the rtp packet, if it >>> gets >>> it >> >> Could you point me how the debug should be conducted? > > Using the asterisk cli turn on debug for the peer and rtp and see what > happens. Match it with the asterisk processes. You have to do this, you can > look at cli or the log files, follow it through to see the rtp packet being > received. Lots of debug advice on google. Asterisk cli did not show anything interesting. I tried pjsip set logger verbose on, but no logs showed anywhere. What am I doing wrong? Marek >> >> Is my suspection that the problem could be caused by nat ip addres >> changing reasonable? How should asterisk handle the situation? > I can’t see anything to support that. Everything is looking normal except > asterisk doesn’t appear to beseeing the rtp packet >> >> Thanks >> >> Marek >> >> >>> >>> Have fun, its all good learning. >>> >>> >>>> On Sun, Sep 5, 2021 at 6:27 PM Marek Greško wrote: >>>> >>>> Hello, >>>> >>>> regarding the ipv6, you see nothing about that it should be some type >>>> of ipv6 tunnelling, because also MTU is lower than expected. You >>>> should not see any ipv6 related communication in the sniff. Phone is >>>> not aware of it. >>>> >>>> The asterisk's static public ip address is 198.51.100.1. >>>> The remote provider's dynamic nat pool is 192.0.2.0/24. By provider we >>>> mean internet provider the remote phones are behind. We are not >>>> complaining about voip provider, we have no problem with that. Only >>>> communication between asterisk and remote phones behind some internet >>>> provider. This is the only conversation to look at. >>>> The phone private address is 192.168.100.235. >>>> >>>> Thanks >>>> >>>> Marek >>>> >>>> >>>> 2021-09-05 1:11 GMT+02:00, Duncan Turnbull : >>>>> >>>>> >>>>>> On 5/09/2021, at 10:21 AM, Marek Greško wrote: >>>>>> >>>>>> Hello, >>>>>> >>>>>> could you please answer my previous question about anonymizing several >>>>>> parameters? I have the data ready, but will post after answer. I have >>>>>> no clue whether I could disclose some important data not deleting >>>>>> them. >>>>>> >>>>>> Regarding sdp, the address will be the internal one, since the phone >>>>>> is behind nat and it is not aware of the nat. The provider's nat >>>>>> device is configured as dump nat, no application tweaking is done. So >>>>>> the asterisk will see the lan address in the sip. >>>>>> >>>>> There are two conversations to look at >>>>> Provider to Asterisk >>>>> Asterisk to Phone >>>>> You need the
Re: [asterisk-users] problems with natted phones
Hello, 2021-09-06 2:51 GMT+02:00, Duncan Turnbull : > Hi Marek > > I didn't understand your setup originally. > > Can you confirm this is correct: > > You provide asterisk for a number of remote phones. I assume they register > to the asterisk > > Asterisk ( 198.51.100.1) <==> Phone Provider ( 192.0.2.0/24 ) <==> Phone ( > 192.168.100.235 ) > > Your call that fail is coming from asterisk to the phone offering G711A, > G729, iLBC, GSM, G723 and rtp on port 18892 Exactly correct. > > Its unclear to me still whether the remote provider has a SIP device in > front of the phones or just a firewall. The user agent for the reply is It is just a firewall. I disabled SIP ALG on it. The nat is performed probably somewhere in the provider's network. I see only ipv6 tunnel to the provider's netwrork. > A540 which I am not familiar with The second phone is Cisco SPA502G. Same problems. > > The call that works shows the Asterisk sending to the internal ip until it > receives rtp from the remote phone from which it learns its address and > port and redirects the rtp to. This is fairly standard I would expect that when asterisk is aware of nat, it does not send the rtp until it receives rtp from other side to learn the port, but OK, no problem to accept the behavior. > > For the case of the call that doesn't work, your asterisk receives the rtp > with the external address but doesn't learn from it. Yes exactly, but I do not undestand why. And why the reboot of the provider's router helps to solve the problem for several days? > > You haven't provided the full call data including the close down of the > call and the registrations would have been helpful too but no matter. > > The question is why your asterisk didn't learn the external address and > port from the received rtp packet > > You can look at your logs with debug to see what decisions its making. You > can see if different rtp ports have different results. > Your phone provider has rtp on 5010 unsuccessfully and 5016 successfully. > Your asterisk uses rtp 13786 successfully and fails when using 18892. Is it > possible your firewall is blocking port 18892 and so asterisk never sees > the returned packet and can't learn from it? It is very unprobable. I see no reason for blocking the port. The problem is asterisk never learns the correct port, so there is nothing to block. > > In any event you should put your debug on and look at your logs in asterisk > to see what it sees and why it doesn't react to the rtp packet, if it gets > it Could you point me how the debug should be conducted? Is my suspection that the problem could be caused by nat ip addres changing reasonable? How should asterisk handle the situation? Thanks Marek > > Have fun, its all good learning. > > > On Sun, Sep 5, 2021 at 6:27 PM Marek Greško wrote: > >> Hello, >> >> regarding the ipv6, you see nothing about that it should be some type >> of ipv6 tunnelling, because also MTU is lower than expected. You >> should not see any ipv6 related communication in the sniff. Phone is >> not aware of it. >> >> The asterisk's static public ip address is 198.51.100.1. >> The remote provider's dynamic nat pool is 192.0.2.0/24. By provider we >> mean internet provider the remote phones are behind. We are not >> complaining about voip provider, we have no problem with that. Only >> communication between asterisk and remote phones behind some internet >> provider. This is the only conversation to look at. >> The phone private address is 192.168.100.235. >> >> Thanks >> >> Marek >> >> >> 2021-09-05 1:11 GMT+02:00, Duncan Turnbull : >> > >> > >> >> On 5/09/2021, at 10:21 AM, Marek Greško wrote: >> >> >> >> Hello, >> >> >> >> could you please answer my previous question about anonymizing several >> >> parameters? I have the data ready, but will post after answer. I have >> >> no clue whether I could disclose some important data not deleting >> >> them. >> >> >> >> Regarding sdp, the address will be the internal one, since the phone >> >> is behind nat and it is not aware of the nat. The provider's nat >> >> device is configured as dump nat, no application tweaking is done. So >> >> the asterisk will see the lan address in the sip. >> >> >> > There are two conversations to look at >> > Provider to Asterisk >> > Asterisk to Phone >> > You need the packet captures of both. >> > >> > Your statements are mixing them up >> > >> > I don’t know what you mean by LAN addr
Re: [asterisk-users] problems with natted phones
Hello, regarding the ipv6, you see nothing about that it should be some type of ipv6 tunnelling, because also MTU is lower than expected. You should not see any ipv6 related communication in the sniff. Phone is not aware of it. The asterisk's static public ip address is 198.51.100.1. The remote provider's dynamic nat pool is 192.0.2.0/24. By provider we mean internet provider the remote phones are behind. We are not complaining about voip provider, we have no problem with that. Only communication between asterisk and remote phones behind some internet provider. This is the only conversation to look at. The phone private address is 192.168.100.235. Thanks Marek 2021-09-05 1:11 GMT+02:00, Duncan Turnbull : > > >> On 5/09/2021, at 10:21 AM, Marek Greško wrote: >> >> Hello, >> >> could you please answer my previous question about anonymizing several >> parameters? I have the data ready, but will post after answer. I have >> no clue whether I could disclose some important data not deleting >> them. >> >> Regarding sdp, the address will be the internal one, since the phone >> is behind nat and it is not aware of the nat. The provider's nat >> device is configured as dump nat, no application tweaking is done. So >> the asterisk will see the lan address in the sip. >> > There are two conversations to look at > Provider to Asterisk > Asterisk to Phone > You need the packet captures of both. > > Your statements are mixing them up > > I don’t know what you mean by LAN address, that’s an ambiguous term. The ip > your asterisk receives from the provider should be the providers external ip > or in the sdp the external address of the media server which may or may not > be the same device > >> In the working scenario it is sending rtp packets to the internal >> address which is wrong, but after receiving cca 5 rtp packets from the >> phone it somehow discovers correct nat ip/port and switches to it. In >> non-working scenario it never switches and still sends to the lan >> address. Strange there is no audio, even one direction. Another >> strange thing is there are 2 phones (different vendors) behind the >> same nat and the problem appearance on them is independent, sometimes >> the first has problem, sometimes the second and sometimes both. >> >> The tcpdumps are made on the asterisk side. I have currently no means >> of capturing on phone side. >> >> Marek >> >> 2021-09-04 23:56 GMT+02:00, Antony Stone >> : >>>> On Saturday 04 September 2021 at 22:13:32, Marek Greško wrote: >>>> >>>> Hello, >>>> >>>> I agree my knowledge of SIP itself is poor, but I have quite well >>>> general tcp/ip understanding. What sip parameters should be >>>> anonymized? How about tag, branch, call-id, cseq values? >>> >>> Show us your packet captures with meaningful addresses (not necessarily >>> accurate ones, but at least unambiguous - see my previous suggestion re >>> RFC5737) and we can help you to understand them and what they mean. >>> >>> >>> Antony. >>> >>> -- >>> Heisenberg, Gödel, and Chomsky walk in to a bar. >>> Heisenberg says, "Clearly this is a joke, but how can we work out if it's >>> funny or not?" >>> Gödel replies, "We can't know that because we're inside the joke." >>> Chomsky says, "Of course it's funny. You're just saying it wrong." >>> >>> 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 >
Re: [asterisk-users] problems with natted phones
Hello, I first tried to communicate to internet provider but without any result. They told me I should find problem on my side. It does not seem the provider is blocking SIP, since it is working for some time and then stops. But I suspect the problem could be the NAT address change. The provider offers only some IPv6 tunnel to his network and the NAT is done probably on other property, I am not sure. But the local provider's router does not possess any ipv4 address on the external interface, only ipv6. I noticed the change now, when anonymizing the tcpdumps. But the working scenario is after reboot, so the ip address change is expected, but I cannot guarantee it does not change in time. Maybe my previous expectation it is caused by asterisk reboot was misleading. Could this be the cause? If yes, are there any means how to overcome it on the asterisk side? I attached the tcpdumps. Do you see something inconsistent in it? By algorithms I meant when the router actively translates the sip payload to change the lan addresses to the natted one. Thanks Marek 2021-09-05 0:40 GMT+02:00, Antony Stone : > On Thursday 08 July 2021 at 20:57:30, Marek Greško wrote: > >> Hello, >> >> I have an asterisk setup using pjsip. Everything used to work >> correctly until one remote site changed internet provider and thier >> router does not support sip protocol algorithms. > > I'm slightly bothered by the word "algorithms" in that comment, but I do > wonder whether it simply means that this is a connectivity provider > (possibly > a mobile phone network?) which actively blocks SIP. > > Some of them (in my experience) do this by blocking UDP port 5060, but > others > are more subtle about it, and (for example) block the authentication > responses > to a Register request, thereby meaning that UDP port 5060 is in general > accessible, but any attempt to Register to it using SIP will fail. > > Have you asked the new Internet connectivity provider whether they support > or > block SIP across their network? > > > Antony > > -- > "Remember: the S in IoT stands for Security." > > - Jan-Piet Mens > >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 21:39:47.710512 IP 192.0.2.1.32260 > 198.51.100.1.5060: SIP 21:39:49.950775 IP 198.51.100.1.5060 > 192.0.2.1.32260: SIP: INVITE sip:999@192.0.2.1:32260 SIP/2.0 INVITE sip:999@192.0.2.1:32260 SIP/2.0 Via: SIP/2.0/UDP 198.51.100.1:5060;rport;branch=z9hG4bKPjf0991750-2e98-4f19-9749-c4eee08a4e37 From: ;tag=e842dd73-cda0-45f5-81a7-a8fcc1472ac9 To: Contact: Call-ID: 603c61b6-5679-452a-aafa-0436f8d8b672 CSeq: 19604 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 18.2.0 Content-Type: application/sdp Content-Length: 376 v=0 o=- 121003081 121003081 IN IP4 198.51.100.1 s=Asterisk c=IN IP4 198.51.100.1 t=0 0 m=audio 13786 RTP/AVP 8 18 97 3 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv 21:39:50.022921 IP 192.0.2.1.32260 > 198.51.100.1.5060: SIP: SIP/2.0 100 Trying SIP/2.0 100 Trying Via: SIP/2.0/UDP 198.51.100.1:5060;rport=5060;branch=z9hG4bKPjf0991750-2e98-4f19-9749-c4eee08a4e37 From: ;tag=e842dd73-cda0-45f5-81a7-a8fcc1472ac9 To: ;tag=1427222172 Call-ID: 603c61b6-5679-452a-aafa-0436f8d8b672 CSeq: 19604 INVITE Contact: User-Agent: A540 IP/42.247.00.000.000 Content-Length: 0 21:39:50.559818 IP 192.0.2.1.32260 > 198.51.100.1.5060: SIP 21:39:50.563295 IP 192.0.2.1.32260 > 198.51.100.1.5060: SIP 21:39:50.568182 IP 192.0.2.1.32260 > 198.51.100.1.5060: SIP: SIP/2.0 180 Ringing SIP/2.0 180 Ringing Via: SIP/2.0/UDP 198.51.100.1:5060;rport=5060;branch=z9hG4bKPjf0991750-2e98-4f19-9749-c4eee08a4e37 From: ;tag=e842dd73-cda0-45f5-81a7-a8fcc1472ac9 To: ;tag=1427222172 Call-ID: 603c61b6-5679-452a-aafa-0436f8d8b672 CSeq: 19604 INVITE Contact: Allow-Events: message-summary, refer, ua-profile, talk, check-sync User-Agent: A540 IP/42.247.00.000.0
Re: [asterisk-users] problems with natted phones
Hello, could you please answer my previous question about anonymizing several parameters? I have the data ready, but will post after answer. I have no clue whether I could disclose some important data not deleting them. Regarding sdp, the address will be the internal one, since the phone is behind nat and it is not aware of the nat. The provider's nat device is configured as dump nat, no application tweaking is done. So the asterisk will see the lan address in the sip. In the working scenario it is sending rtp packets to the internal address which is wrong, but after receiving cca 5 rtp packets from the phone it somehow discovers correct nat ip/port and switches to it. In non-working scenario it never switches and still sends to the lan address. Strange there is no audio, even one direction. Another strange thing is there are 2 phones (different vendors) behind the same nat and the problem appearance on them is independent, sometimes the first has problem, sometimes the second and sometimes both. The tcpdumps are made on the asterisk side. I have currently no means of capturing on phone side. Marek 2021-09-04 23:56 GMT+02:00, Antony Stone : > On Saturday 04 September 2021 at 22:13:32, Marek Greško wrote: > >> Hello, >> >> I agree my knowledge of SIP itself is poor, but I have quite well >> general tcp/ip understanding. What sip parameters should be >> anonymized? How about tag, branch, call-id, cseq values? > > Show us your packet captures with meaningful addresses (not necessarily > accurate ones, but at least unambiguous - see my previous suggestion re > RFC5737) and we can help you to understand them and what they mean. > > > Antony. > > -- > Heisenberg, Gödel, and Chomsky walk in to a bar. > Heisenberg says, "Clearly this is a joke, but how can we work out if it's > funny or not?" > Gödel replies, "We can't know that because we're inside the joke." > Chomsky says, "Of course it's funny. You're just saying it wrong." > >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
Re: [asterisk-users] problems with natted phones
Hello, I agree my knowledge of SIP itself is poor, but I have quite well general tcp/ip understanding. What sip parameters should be anonymized? How about tag, branch, call-id, cseq values? Thanks Marek 2021-09-04 12:36 GMT+02:00, Duncan Turnbull : > > >> On 4/09/2021, at 8:55 PM, Marek Greško wrote: >> >> Ok, >> >> let substitute lan for 192.168.100.235, provider with 192.0.2.1 and >> asterisk with 198.51.100.1. > > Can you provide the previous packet details with these addresses filled in >> >> In the working scenario understand that asterisk is not aware of the >> providers ip address > If the call goes provider - asterisk - phone then asterisk is absolutely > aware of the provider ip. I think you need to get more familiar with sip and > rtp > >> 192.0.2.1 in the sip protocol, and it should pick >> it from the network layer. It is harder to calcutale port, so it >> should probably listen for incoming rtp stream? > > The sdp in the sip packet tells the rtp ip and port to connect to >> Until then it is just >> sending to private address? But I thing it is futile, since it is >> known from the sip protocol there is nat involved and thus the packets >> are destined to nowhere. > > You need to realise that this works normally everyday all over the place so > what you are imagining is incorrect >> >> But I still cannot imagine what goes wrong in non working scenario and >> how the asterisk reboot (not every one and not sure this is the real >> trigger). The sip communication seems identical to me. The provider's >> router does not touch SIP now as observed after disabling SIP ALG. > > It is very unclear as to how you are justifying these statements. You don’t > yet understand how sip and call setup with media works. If you provide the > whole sip packet capture with the substituted ips it should be easier to > point out where the error is > > You need to be really clear on what’s ip > is what and where the conversations are captured > > It will become clear once you provide all the details > > >> >> Thanks >> >> Marek >> >> 2021-09-04 0:40 GMT+02:00, Antony Stone >> : >>> On Saturday 04 September 2021 at 00:34:49, Duncan Turnbull wrote: >>> >>>>>> On 4/09/2021, at 7:53 AM, Marek Greško wrote: >>>>>> >>>>>> So you suspect something is messing up SIP protocol? Maybe the phone >>>>>> itself is not working properly. The phone itself is not aware of the >>>>>> internet address, so is sending lan private address in the sip >>>>>> protocol. >>>> >>>> I doubt it’s the phone. You need to check your data again and ideally >>>> explain what you mean by the names you have substituted for the ip >>>> addresses >>> >>> My advice (regarding the IP addresses) is: >>> >>> - where you have https://tools.ietf.org/html/rfc1918 addresses, leave >>> them >>> as >>> they are - you're not giving away any sensitive information by telling us >>> about your internal addresses which can't be routed over the Internet >>> >>> - where you have public addresses and would prefer not to reveal what >>> these >>> are, substitute with https://tools.ietf.org/html/rfc5737 addresses >>> instead. >>> >>> - always ensure that you substitute address A in the same way each time, >>> and >>> address B, etc. >>> >>> >>> Antony. >>> >>> -- >>> You can spend the whole of your life trying to be popular, >>> but at the end of the day the size of the crowd at your funeral >>> will be largely dictated by the weather. >>> >>> - Frank Skinner >>> >>> 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] problems with natted phones
Ok, let substitute lan for 192.168.100.235, provider with 192.0.2.1 and asterisk with 198.51.100.1. In the working scenario understand that asterisk is not aware of the providers ip address 192.0.2.1 in the sip protocol, and it should pick it from the network layer. It is harder to calcutale port, so it should probably listen for incoming rtp stream? Until then it is just sending to private address? But I thing it is futile, since it is known from the sip protocol there is nat involved and thus the packets are destined to nowhere. But I still cannot imagine what goes wrong in non working scenario and how the asterisk reboot (not every one and not sure this is the real trigger). The sip communication seems identical to me. The provider's router does not touch SIP now as observed after disabling SIP ALG. Thanks Marek 2021-09-04 0:40 GMT+02:00, Antony Stone : > On Saturday 04 September 2021 at 00:34:49, Duncan Turnbull wrote: > >> > On 4/09/2021, at 7:53 AM, Marek Greško wrote: >> > >> > So you suspect something is messing up SIP protocol? Maybe the phone >> > itself is not working properly. The phone itself is not aware of the >> > internet address, so is sending lan private address in the sip >> > protocol. >> >> I doubt it’s the phone. You need to check your data again and ideally >> explain what you mean by the names you have substituted for the ip >> addresses > > My advice (regarding the IP addresses) is: > > - where you have https://tools.ietf.org/html/rfc1918 addresses, leave them > as > they are - you're not giving away any sensitive information by telling us > about your internal addresses which can't be routed over the Internet > > - where you have public addresses and would prefer not to reveal what these > are, substitute with https://tools.ietf.org/html/rfc5737 addresses instead. > > - always ensure that you substitute address A in the same way each time, > and > address B, etc. > > > Antony. > > -- > You can spend the whole of your life trying to be popular, > but at the end of the day the size of the crowd at your funeral > will be largely dictated by the weather. > > - Frank Skinner > >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
Re: [asterisk-users] problems with natted phones
So you suspect something is messing up SIP protocol? Maybe the phone itself is not working properly. The phone itself is not aware of the internet address, so is sending lan private address in the sip protocol. I would expect asterisk itself is pairing the provider address with the lan address. I was asked to disable all the SIP ALG on the provider's router in the previous discussion. And it made a big improvement in the experience. Marek 2021-09-03 12:19 GMT+02:00, Duncan Turnbull : > On Fri, Sep 3, 2021 at 8:47 PM Marek Greško wrote: > >> Hello, >> >> I looked into tcpdumps. When problem starts (after some asterisk >> reboot) the call looks like this: >> >> provider:25298 -> asterisk:5060 >> SIP: SIP/2.0 200 OK >> Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=... >> From: ;tag=... >> To: ;tag=... >> Call-ID: ... >> CSeq: ... INVITE >> Contact: >> Supported: replaces >> Allow-Events: message-sumary, refer, ua-profile, talk, check-sync >> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS. INFO, SUBSCRIBE, NOTIFY, REFER, >> UPDATE >> Content-Type: application/sdp >> Content-Length: ... >> >> v=0 >> o=... 5010 ... IN IP4 lan >> s=Mapping >> > This bit here tells where the rtp has to go to. I don't think you want it > to be IP4 lan. It would be a lot more helpful if you had the ip address but > the use of the word LAN suggests its a private IP which asterisk is not > going to be able to route to > > >> c=IN IP4 lan >> t=0 0 >> m=audio 5010 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=sendrcv >> a=ptime:20 >> >> asterisk:5060 -> provider:25298 >> Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=... >> From: ;tag=... >> To: ;tag=... >> Call-ID: ... >> CSeq: ... ACK >> Max-Forwards: 70 >> User-Agent: Asterisk PBX 18.2.0 >> Content-Length: 0 >> >> Then I see RTP packets: >> asterisk:18892 -> lan:5010 >> provider:25420 -> asterisk:18892 >> > As above for RTP to work they have to go to/from the end points. Asterisk > is sending to 18892 instead of the provider 25420 > > Why is your provider sending you an sdp with rtp with a private ip address? > Or are they sending the right address and your ALG or something else is > changing it? Ask your provider what they are sending you? Then find out > who/what is messing up the SDP > > >> >> I hear no audio. I heard stream towards the asterisk prior to SIP ALG >> disabling. Now silence both directions. It should not be a codec >> problem. After providers router reboot I can hear both directions but >> it still seems weird: >> >> provider:32260 -> asterisk:5060 >> SIP: SIP/2.0 200 OK >> Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=... >> From: ;tag=... >> To: ;tag=... >> Call-ID: ... >> CSeq: ... INVITE >> Contact: >> Supported: replaces >> Allow-Events: message-sumary, refer, ua-profile, talk, check-sync >> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS. INFO, SUBSCRIBE, NOTIFY, REFER, >> UPDATE >> Content-Type: application/sdp >> Content-Length: ... >> >> v=0 >> o=... 5016 ... IN IP4 lan >> s=Mapping >> c=IN IP4 lan >> t=0 0 >> m=audio 5016 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=sendrcv >> a=ptime:20 >> >> asterisk:5060 -> provider:32260 >> Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=... >> From: ;tag=... >> To: ;tag=... >> Call-ID: ... >> CSeq: ... ACK >> Max-Forwards: 70 >> User-Agent: Asterisk PBX 18.2.0 >> Content-Length: 0 >> >> Then I see several RTP packets: >> asterisk:13786 -> lan:5016 >> provider:32327 -> asterisk:13786 >> for a while and the suddenly >> asterisk:13786 -> provider:32327 >> provider:32327 -> asterisk:13786 >> >> The user experience for that scenario is OK. >> >> I suspect some configuration error on asterisk side, since also for >> working scenario I see RTP packets to the lan. But I cannot figure out >> what it is. When I was using another provider which had working SIP >> ALG I had no problem even without nat configuration on the asterisk >> side. >> >> The experience is clearly better after disabling SIP ALG, but we still >> face problems after asterisk side reboots. >> >> Could you point me for what should I look in the asterisk >> configuration? And why the problems are gone after provider's router >
Re: [asterisk-users] problems with natted phones
Hello, I looked into tcpdumps. When problem starts (after some asterisk reboot) the call looks like this: provider:25298 -> asterisk:5060 SIP: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=... From: ;tag=... To: ;tag=... Call-ID: ... CSeq: ... INVITE Contact: Supported: replaces Allow-Events: message-sumary, refer, ua-profile, talk, check-sync Allow: INVITE, ACK, CANCEL, BYE, OPTIONS. INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE Content-Type: application/sdp Content-Length: ... v=0 o=... 5010 ... IN IP4 lan s=Mapping c=IN IP4 lan t=0 0 m=audio 5010 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrcv a=ptime:20 asterisk:5060 -> provider:25298 Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=... From: ;tag=... To: ;tag=... Call-ID: ... CSeq: ... ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.2.0 Content-Length: 0 Then I see RTP packets: asterisk:18892 -> lan:5010 provider:25420 -> asterisk:18892 I hear no audio. I heard stream towards the asterisk prior to SIP ALG disabling. Now silence both directions. It should not be a codec problem. After providers router reboot I can hear both directions but it still seems weird: provider:32260 -> asterisk:5060 SIP: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=... From: ;tag=... To: ;tag=... Call-ID: ... CSeq: ... INVITE Contact: Supported: replaces Allow-Events: message-sumary, refer, ua-profile, talk, check-sync Allow: INVITE, ACK, CANCEL, BYE, OPTIONS. INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE Content-Type: application/sdp Content-Length: ... v=0 o=... 5016 ... IN IP4 lan s=Mapping c=IN IP4 lan t=0 0 m=audio 5016 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrcv a=ptime:20 asterisk:5060 -> provider:32260 Via: SIP/2.0/UDP asterisk:5060;rport=5060;branch=... From: ;tag=... To: ;tag=... Call-ID: ... CSeq: ... ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.2.0 Content-Length: 0 Then I see several RTP packets: asterisk:13786 -> lan:5016 provider:32327 -> asterisk:13786 for a while and the suddenly asterisk:13786 -> provider:32327 provider:32327 -> asterisk:13786 The user experience for that scenario is OK. I suspect some configuration error on asterisk side, since also for working scenario I see RTP packets to the lan. But I cannot figure out what it is. When I was using another provider which had working SIP ALG I had no problem even without nat configuration on the asterisk side. The experience is clearly better after disabling SIP ALG, but we still face problems after asterisk side reboots. Could you point me for what should I look in the asterisk configuration? And why the problems are gone after provider's router reboot? Thanks Marek 2021-08-13 15:31 GMT+02:00, Duncan Turnbull : > >>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 disab
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? 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
Re: [asterisk-users] problems with natted phones
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
Re: [asterisk-users] problems with natted phones
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
Re: [asterisk-users] problems with natted phones
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
Re: [asterisk-users] problems with natted phones
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
Re: [asterisk-users] problems with natted phones
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
Re: [asterisk-users] problems with natted phones
To be more specific I was on the https://wiki.asterisk.org/wiki/display/AST/Getting+Started already, but I assume all the additional transport parameters are relevant only when asterisk itself is behind nat. Is not it true? Marek 2021-07-09 8:47 GMT+02:00, Marek Greško : > Hello, > > yes SIP ALG are anbled on the router. Should I disable? > > Transport config looks like that: > > [transport-udp] > type = transport > protocol = udp > bind = 0.0.0.0 > domain = mydomain.com > > Asterisk itself is not natted. > > Marek > > > 2021-07-08 21:14 GMT+02:00, Michael L. Young : >> El jue, 8 de jul. de 2021 a la(s) 14:58, Marek Greško >> (mgres...@gmail.com) >> escribió: >> >> >>> The asterisk is connected to the internet with public static IP address. >>> >>> The pjsip config contains: >>> >>> >> What does your transport config look like? >> >> Take a look at this wiki page: >> https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT >> >> -- >> Michael L. Young >> (elguero) >> > -- _ -- 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, yes SIP ALG are anbled on the router. Should I disable? Transport config looks like that: [transport-udp] type = transport protocol = udp bind = 0.0.0.0 domain = mydomain.com Asterisk itself is not natted. Marek 2021-07-08 21:14 GMT+02:00, Michael L. Young : > El jue, 8 de jul. de 2021 a la(s) 14:58, Marek Greško (mgres...@gmail.com) > escribió: > > >> The asterisk is connected to the internet with public static IP address. >> >> The pjsip config contains: >> >> > What does your transport config look like? > > Take a look at this wiki page: > https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT > > -- > Michael L. Young > (elguero) > -- _ -- 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] problems with natted phones
Hello, I have an asterisk setup using pjsip. Everything used to work correctly until one remote site changed internet provider and thier router does not support sip protocol algorithms. It works for some time, but then suddenly audio stops working both directions. When this happens I see RTP responses going out to the local address of the natted phone, not to the natted address. The problem appears for the phones independently. The asterisk is connected to the internet with public static IP address. The pjsip config contains: [aor] type=aor qualify_frequency = 60 max_contacts=1 remove_existing = yes [endpoint] type = endpoint context = internal dtmf_mode = rfc4733 disallow = all allow = alaw allow = ilbc allow = g729 allow = gsm allow = g723 direct_media = no allow_subscribe = yes subscribe_context = blf rewrite_contact = yes rtp_symmetric = yes force_rport = yes Am I missing something? Why the communication breaks suddenly? Thanks Marek -- _ -- 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] pjsip presence on Cisco SPA525G2 with SPA500DS
Hello, could somebody drive me how could I make run presence reporting by BLF feature on the Cisco SPA525G2 with SPA500DS on asterisk with pjsip stack? I am not able to configure asterisk side. When I run pjsip show subscriptions inbound I see all subscriptions as dialog. Which as of my understanding is not sufficient. Moreover I suspect even I configure asterisk side successfully, the SPA525G2 phones will not support it. Could somebody confirm if it is working or not possible to achieve it using these phones? Thanks. Marek -- _ -- 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] Caller cannot blind transfer
Hello,
I am not sure whether this was correct solution, but I overcommed the
issue by defining context [gosub-stdexten] and including the same
definition of dialplan
exten => xx,1,Dial(PJSIP/xx,30,TtrKk)
exten => xy,1,GoSub(gosub-stdexten,s,1(xy,PJSIP/xy,30,TtrKk))
in it.
It seems the procedure runs in its own context gosub-stdexten. If you
are aware of any better solution I would be glad to here about.
Marek
2021-01-05 18:32 GMT+01:00, Marek Greško :
> Hello,
>
> I am unable to figure out why I am not able to blind transfer when I
> am the caller and I call the extension defined by gosub.
>
> When running asterisk -rvvv I can see:
>
> -- Channel PJSIP/-0009: Dialed ' number>@gosub-stdexten' does not exist.
>
> It is evident there has been added some weird context after the
> extension number. The gosub-stdexten is a name of Gosub Procedure. Why
> it used it as a context? Where is the context name read from?
>
> The extensions are defined as follows:
> exten => xx,1,Dial(PJSIP/xx,30,TtrKk)
> exten => xy,1,GoSub(gosub-stdexten,s,1(xy,PJSIP/xy,30,TtrKk))
>
> xx is a caller, xy is a callee
>
> The procedure gosub-stdexten itself looks like this:
>
>
> [gosub-stdexten]
> ;
> ; Standard extension macro:
> ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
> ; ${ARG2} - Device(s) to ring
> ; ${ARG3} - How long
> ; ${ARG4} - Options
> ; Retrieve the Call Forward number if available.
> exten => s,1,Set(CFIM=${DB(CFIM/${ARG1})})
> ;
> ; Dial the appropriate number depending on whether the Call Forward
> exten => s,n,GotoIf($["${vmbox}"!=""]?s-NoCFIM,1)
> exten => s,n,Set(_vmbox=${ARG1})
> exten => s,n,GotoIf($["${CFIM}"!=""]?s-CFIM,1:s-NoCFIM,1)
> ;
> ; Pass call to VoiceMail with the appropriate greeting.
> ;exten => s,n,GosubIf($[${DIALSTATUS}=BUSY]?s-BUSY,1:s-NOANSWER,1)
> ; Hangup.
> exten => s,n,Hangup()
> ;
> ; Dial Call Forward number & return.
> exten => s-CFIM,1,Dial(Local/${CFIM}@,${ARG3},${ARG4})
> exten => s-CFIM,n,GosubIf($[${DIALSTATUS}=BUSY]?s-BUSY,1:s-NOANSWER,1)
> exten => s-CFIM,n,Hangup()
> ; Dial actual extension & return.
> exten => s-NoCFIM,1,Dial(${ARG2},${ARG3},${ARG4})
> exten => s-NoCFIM,n,GosubIf($[${DIALSTATUS}=BUSY]?s-BUSY,1:s-NOANSWER,1)
> exten => s-NoCFIM,n,Hangup()
> ;
> ; Unavailable voicemail message if there is no answer.
> exten => s-NOANSWER,1,GotoIf($["${vmbox}"=""]?3:2)
> exten => s-NOANSWER,2,VoiceMail(${vmbox}@|u)
> exten => s-NOANSWER,3,Return()
> ; Busy voicemail message for any DIALSTATUS other than NOANSWER (or
> ANSWER).
> exten => s-BUSY,1,GotoIf($["${vmbox}"=""]?3:2)
> exten => s-BUSY,2,VoiceMail(${vmbox}@|b)
> exten => s-BUSY,3,Return()
>
>
> How could I fix it? Should I forward the original context somehow into the
> exten => s-NoCFIM,1,Dial(${ARG2},${ARG3},${ARG4})
> ? And also maybe here
> exten => s-CFIM,1,Dial(Local/${CFIM}@,${ARG3},${ARG4})
> ?
>
> Thanks
>
> Marek
>
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[asterisk-users] Caller cannot blind transfer
Hello,
I am unable to figure out why I am not able to blind transfer when I
am the caller and I call the extension defined by gosub.
When running asterisk -rvvv I can see:
-- Channel PJSIP/-0009: Dialed '@gosub-stdexten' does not exist.
It is evident there has been added some weird context after the
extension number. The gosub-stdexten is a name of Gosub Procedure. Why
it used it as a context? Where is the context name read from?
The extensions are defined as follows:
exten => xx,1,Dial(PJSIP/xx,30,TtrKk)
exten => xy,1,GoSub(gosub-stdexten,s,1(xy,PJSIP/xy,30,TtrKk))
xx is a caller, xy is a callee
The procedure gosub-stdexten itself looks like this:
[gosub-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - How long
; ${ARG4} - Options
; Retrieve the Call Forward number if available.
exten => s,1,Set(CFIM=${DB(CFIM/${ARG1})})
;
; Dial the appropriate number depending on whether the Call Forward
exten => s,n,GotoIf($["${vmbox}"!=""]?s-NoCFIM,1)
exten => s,n,Set(_vmbox=${ARG1})
exten => s,n,GotoIf($["${CFIM}"!=""]?s-CFIM,1:s-NoCFIM,1)
;
; Pass call to VoiceMail with the appropriate greeting.
;exten => s,n,GosubIf($[${DIALSTATUS}=BUSY]?s-BUSY,1:s-NOANSWER,1)
; Hangup.
exten => s,n,Hangup()
;
; Dial Call Forward number & return.
exten => s-CFIM,1,Dial(Local/${CFIM}@,${ARG3},${ARG4})
exten => s-CFIM,n,GosubIf($[${DIALSTATUS}=BUSY]?s-BUSY,1:s-NOANSWER,1)
exten => s-CFIM,n,Hangup()
; Dial actual extension & return.
exten => s-NoCFIM,1,Dial(${ARG2},${ARG3},${ARG4})
exten => s-NoCFIM,n,GosubIf($[${DIALSTATUS}=BUSY]?s-BUSY,1:s-NOANSWER,1)
exten => s-NoCFIM,n,Hangup()
;
; Unavailable voicemail message if there is no answer.
exten => s-NOANSWER,1,GotoIf($["${vmbox}"=""]?3:2)
exten => s-NOANSWER,2,VoiceMail(${vmbox}@|u)
exten => s-NOANSWER,3,Return()
; Busy voicemail message for any DIALSTATUS other than NOANSWER (or ANSWER).
exten => s-BUSY,1,GotoIf($["${vmbox}"=""]?3:2)
exten => s-BUSY,2,VoiceMail(${vmbox}@|b)
exten => s-BUSY,3,Return()
How could I fix it? Should I forward the original context somehow into the
exten => s-NoCFIM,1,Dial(${ARG2},${ARG3},${ARG4})
? And also maybe here
exten => s-CFIM,1,Dial(Local/${CFIM}@,${ARG3},${ARG4})
?
Thanks
Marek
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Re: [asterisk-users] Sollicitated MWI not working
Hello,
thanks for tip. You were absolutely right. I did not realize the
voicemail context is defined in voicemail.conf. After adding the
context to the mailboxes configuration in endpoint it started working
correctly.
Morever using tcpdump I found out that defining mailboxes in both aor
and endpoint you get 500 as a response (but sitll working). It should
be defined in only one of them. Strange it is working also for
unsolicited if defined only in aor.
I did not get solicited working in either hardware phone or twinkle.
The unsolicited one is working in both.
Marek
2021-01-02 14:35 GMT+01:00, Joshua C. Colp :
> On Sat, Jan 2, 2021 at 5:36 AM Marek Greško wrote:
>
>> Hello,
>>
>> I configured MWI with pjsip.
>>
>> The aors section contains:
>>
>> mailboxes = 101@
>>
>> The endpoint section contains:
>>
>> context = internal
>> mailboxes = 101@
>>
>> The dialplan leaves the voicemail by:
>> exten => s-NOANSWER,2,VoiceMail(${vmbox}@|u)
>> or:
>> exten => s-BUSY,2,VoiceMail(${vmbox}@|b)
>>
>
> You need to specify a context in the "mailboxes" lines, in the form of
> 101@context. The context would depend on what you have configured in
> voicemail.conf. As you have not provided that, I can not say what with
> certainty. Specifying the context should resolve your issue with
> unsolicited MWI taking time to update.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
>
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[asterisk-users] Sollicitated MWI not working
Hello,
I configured MWI with pjsip.
The aors section contains:
mailboxes = 101@
The endpoint section contains:
context = internal
mailboxes = 101@
The dialplan leaves the voicemail by:
exten => s-NOANSWER,2,VoiceMail(${vmbox}@|u)
or:
exten => s-BUSY,2,VoiceMail(${vmbox}@|b)
When I configure clients to use unsollicitated MWI, it works, but
there are quite long delays in status display. So I decided to try
sollicitated MWI, but I am reaching some difficulties.
The twinkle client simply freezes on configuration. By the log I can
see it receives 401 on subscribe and then freezes. The SPA525G2 phone
automatically disables Message Waiting feature.
I tried selectively disabling mailboxes commands in both endpoint
(unsolliciteted stopped to work) and aors section but none of them
worked. I tried to change mailboxes = 101@ to mailboxes = 101 but also
without success.
I am not sure about both asterisk and client configuration.
At asterisk side is it allowed to have mailboxes allowed in both
endpoint and aor section? What is the connert mailbox format? Should
it be specified with @ or is it ok without it? Should the context
after the @ be specified? If so whoch one? The same as context command
in the endpoint? So should I change it to 101@internal? Since the
unsollicitated MWI is working I expect I could leave it as is. Am I
right?
At the phone side, what should be set up? I tried to set Message
Waiting to yes, setting my Voicemail Server to the same name as sip
registry server and I tried mailbox name without @ and with @. Both
without success.
Could you give me some advice how should I configure it correctly, please?
Will the sollicitated MWI help me to evercome the indication delay problem?
Thanks
Marek
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Re: [asterisk-users] Voice broken during calls (again...)
I interchanged LAN and LTE in the sentence. Do you have some kind of NAT in fron of asterisk? Or is your asterisk having public IP? Could you share sip.conf (without passwords)? One LAN client, one LTE and general section. Marek 2020-06-23 16:29 GMT+02:00, Luca Bertoncello : > Am 23.06.2020 16:22, schrieb Marek Greško: >> It seems your problems lie in something other. Most probably it is not >> mtu problem. All my suspections are contradicted. If it is true you >> have inter vlan voice quality problems, it is definitely something >> different. Formerly I assumed you were trying only LTE vs LAN using >> internet. > > I'm not sure what you mean with the last sentence... > I tried to connect to my Asterisk via LAN or via DSL (either via LTE or > other DSL). > Then I noticed that if I call another peer in same network (= both peers > via DSL or both peers in the same VLAN), the quality is very good, > otherwise is very poor. > > But why should Asterisk have problem if the peers are in different > networks it's for me a really big mistery... > > This evening I'll try to capture the pakets in a call between two peers > connected to Asterisk via LTE, two peers connected in the same LAN and a > peer connected via LTE and the other in LAN, then maybe it's possible to > find the problem... > > But if you have any other idea, I'm very happy to hear it! ;) > > 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 -- _ -- 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 (again...)
It seems your problems lie in something other. Most probably it is not mtu problem. All my suspections are contradicted. If it is true you have inter vlan voice quality problems, it is definitely something different. Formerly I assumed you were trying only LTE vs LAN using internet. Marek 2020-06-23 15:50 GMT+02:00, Luca Bertoncello : > Am 23.06.2020 15:43, schrieb Marek Greško: > > Hi > >>> Do you mean "my Linux-Box ignores ICMP packet unreachable" or >>> "Deutsche >>> Telekom ignores them"? >> >> I meant DT, but this was a speculation. I did not say they do. I >> consider it highly improbable. Then I was asking whether you do. As >> per configuration you sent you are not blocking icmp type 3 so this >> should not be an issue. > > OK, so this should not be the problem... > What can we check now? > If you want, I can send my iptables-script. It is possible, that I have > there an error causing this behaviour... > > Maybe someone in the list is an expert with iptables and can check it? > I know this program, but I'm not really an expert... > > 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 -- _ -- 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 (again...)
2020-06-23 15:02 GMT+02:00, Luca Bertoncello : > Am 23.06.2020 14:49, schrieb Marek Greško: > > Hi Marek, > >> this could be ip address of the different interface on the same box. I >> think it works like expected. The only exception would be if the sip >> peer ignores the icmp packet unreachable. But I doubt this is the > > Do you mean "my Linux-Box ignores ICMP packet unreachable" or "Deutsche > Telekom ignores them"? I meant DT, but this was a speculation. I did not say they do. I consider it highly improbable. Then I was asking whether you do. As per configuration you sent you are not blocking icmp type 3 so this should not be an issue. > >> case. Anyway you get problems also when calling to LTE phone without >> using sip provider. > > I have problem calling someone outside my networks and I have problem if > the peers are in different networks... > >> Let first concentrate on these calls LTE to LAN. Are you sure you do >> not block incoming icmp unreachables? At least verify type 3 subtype 4 >> is enabled. If it is, I have no clue what is going on. > > Well, I limit incoming ICMP packets and I block some hosts (known > crackers)... > If you think, I can send you the script I use (with iptables) to manage > my firewall, so you can check it... > The only entries I have, having something to do with ICMP, are: > > -- > /bin/echo -n "Disable ICMP Redirect acceptance..." > for f in /proc/sys/net/ipv4/conf/*/accept_redirects; do >/bin/echo 0 > $f > done > /bin/echo "done." > /sbin/iptables -A INPUT -i dsl0 -p icmp --icmp-type echo-request -m > limit --limit 6/m --limit-burst 5 -j ACCEPT > /sbin/iptables -A FORWARD -o dsl0 -p icmp -j ACCEPT > -- > > and of course other rules to allow ICMP pakets in the internal > networks... > > 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 -- _ -- 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 (again...)
Hello, this could be ip address of the different interface on the same box. I think it works like expected. The only exception would be if the sip peer ignores the icmp packet unreachable. But I doubt this is the case. Anyway you get problems also when calling to LTE phone without using sip provider. Let first concentrate on these calls LTE to LAN. Are you sure you do not block incoming icmp unreachables? At least verify type 3 subtype 4 is enabled. If it is, I have no clue what is going on. Marek Marek 2020-06-23 10:11 GMT+02:00, Luca Bertoncello : > Am 23.06.2020 10:07, schrieb Marek Greško: > > Hi > >> this is a correct response: >> >> From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set >> (mtu = 1492) >> >> So PMTU discovery is working. No problem here. You got correct message >> to lower the packet size from 62.156.246.57. This is probably the last >> hop before your site. > > No, the last hop is 62.156.246.65: > > lucabert@ns:~$ mtr -4nr bpi.d.lucabert.com > Start: Tue Jun 23 10:10:16 2020 > HOST: ns.lucabert.de Loss% Snt Last Avg Best Wrst > StDev >1.|-- 185.242.112.1 0.0%100.4 1.1 0.3 4.4 > 1.2 >2.|-- 84.200.230.82 0.0%100.8 0.7 0.5 0.8 > 0.0 >3.|-- 87.190.233.113 0.0%101.6 1.7 1.4 2.5 > 0.0 >4.|-- 217.5.82.940.0%107.9 7.6 7.4 7.9 > 0.0 >5.|-- 217.5.82.940.0%107.7 7.5 7.2 7.7 > 0.0 >6.|-- 62.156.246.49 0.0%107.4 7.4 7.3 7.4 > 0.0 >7.|-- 62.156.246.65 0.0%107.6 7.6 7.4 7.8 > 0.0 >8.|-- 93.241.91.232 0.0%10 21.4 21.9 21.4 24.3 > 0.7 > > Don't know where this 62.156.246.57 comes... :( > > Everyway: you think, my network works as expected? At least the part > using DSL? > Any idea, where could be the problem? > > 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 -- _ -- 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 (again...)
Hello, this is a correct response: From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set (mtu = 1492) So PMTU discovery is working. No problem here. You got correct message to lower the packet size from 62.156.246.57. This is probably the last hop before your site. Marek 2020-06-23 9:40 GMT+02:00, Luca Bertoncello : > Am 23.06.2020 09:28, schrieb Marek Greško: > > Hi > >> if you need clampmss then it is highly probable there is a PMTU >> discovery problem. The clampmss does not work for UDP. > > Is there a way to check if I have this problem? > >> I probably counted the size incorrectly. So you are able to ping with >> size 1464 and not with 1466. How about trying same ping sizes from the >> internet towards your site? I mean trying to ping from sites with >> higher MTU than yours without lower MTU links in the path. > > lucabert@ns:~$ ping -4 -M do -s 1465 bpi.d.lucabert.com > PING bpi.d.lucabert.com (93.241.91.232) 1465(1493) bytes of data. > From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set > (mtu = 1492) > ping: local error: Message too long, mtu=1492 > ping: local error: Message too long, mtu=1492 > ping: local error: Message too long, mtu=1492 > ^C > --- bpi.d.lucabert.com ping statistics --- > 4 packets transmitted, 0 received, +4 errors, 100% packet loss, time > 3965ms > pipe 2 > > With paket size of 1464 it works... > >> You know MTU is a size of l2 frame, so using ipv6 you are able to use >> higher payload sizes because of ip header size. > > OK, 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 -- _ -- 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 (again...)
Hello, if you need clampmss then it is highly probable there is a PMTU discovery problem. The clampmss does not work for UDP. I probably counted the size incorrectly. So you are able to ping with size 1464 and not with 1466. How about trying same ping sizes from the internet towards your site? I mean trying to ping from sites with higher MTU than yours without lower MTU links in the path. You know MTU is a size of l2 frame, so using ipv6 you are able to use higher payload sizes because of ip header size. Marek 2020-06-23 9:06 GMT+02:00, Luca Bertoncello : > Am 23.06.2020 08:43, schrieb Luca Bertoncello: > > And another thing, I discovered right now... > >> Could you suggest me something to restrict the problem? >> Currently, I think the problem can be: >> >> 1) on Asterisk >> 2) on my Gateway/Firewall > > A couple of years ago I added this entry in my firewall: > > /sbin/iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS > --clamp-mss-to-pmtu > > since I had the problem downloading data from an Internet site using my > tablet. > I found this site explaining that: > > https://lartc.org/howto/lartc.cookbook.mtu-mss.html > > I really forgot this entry, but now I checked all entries in my > Firewall, and I see it, with my remark... > Now, the last line of the HowTo: > > > # iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --set-mss > 128 > > This sets the MSS of passing SYN packets to 128. Use this if you have > VoIP with tiny packets, and huge http packets which are causing chopping > in your voice calls. > > > Could it be the problem? Right now I'm not at home, so I cannot test it, > but maybe I can add an entry like: > > iptables -A FORWARD -p tcp -m multiport --ports 5060, SIP> --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128 > > and change the previous entry like: > > iptables -A FORWARD -p tcp -i intlan0 --tcp-flags SYN,RST SYN -j TCPMSS > --clamp-mss-to-pmtu > > to limit the behaviour on the internal LAN... > > Your opinion? > > 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 -- _ -- 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 (again...)
Hello,
there is no need to change canreinvite for provider configuration.
Do not change MTU. Probably there will be another problem. I expect
packet size 1466 would pass and higher will have the same result. It
would be interesting to make the same test from the outside towards
your asterisk with size 2 bytes larger the highest you are able to
ping.
Marek
2020-06-22 22:26 GMT+02:00, Luca Bertoncello :
> Am 22.06.2020 um 22:12 schrieb Marek Greško:
>
> Hi Marek
>
>> Would you mind repeating the test with canreinvite=no set for all you
>> phones and mobile phones?
>
> All my peers have already canreinvite=no...
> I only have canreinvite=yes on the SIP configuration on the Telekom part:
>
> [pbxluca]
> type=peer
> defaultuser=11...@t-online.de
> secret= xx
> dtmfmode=rfc2833
> host=tel.t-online.de
> context=luca_incoming
> outboundproxy=tel.t-online.de
> port=5060
> fromuser=035
> fromdomain=tel.t-online.de
> usereqphone=yes
> canreinvite=yes
> insecure=port,invite
> nat=no
> qualify=yes
> qualifyfreq=600
> disallow=all
> allow=alaw
> allow=ulaw
>
> Should I change canreinvite=no there?
>
>> What is your upload bitrate? Is it guaranteed?
>
> Currently 12Mbps. Guaranteed should be about 10Mbps...
>
>> I would try also to test the PMTU:
>>
>> Try:
>>
>> ping -M do -s 2000 ${ip address of the sip server}
>>
>> You should receive icmp asking for lowering the packet size.
>
> root@bpi:/etc/asterisk# ping -M do -s 2000 tel.t-online.de
> PING tel.t-online.de (217.0.128.133) 2000(2028) bytes of data.
> ping: local error: Message too long, mtu=1492
> ping: local error: Message too long, mtu=1492
> ping: local error: Message too long, mtu=1492
> ping: local error: Message too long, mtu=1492
> ping: local error: Message too long, mtu=1492
> ping: local error: Message too long, mtu=1492
> ^C
> --- tel.t-online.de ping statistics ---
> 6 packets transmitted, 0 received, +6 errors, 100% packet loss, time 5103ms
>
> Mmmm... it seems not good, isn't it?
>
> For information, here the output of ifconfig:
>
> dsl0: flags=4305 mtu 1492
> inet 93.241.x.y netmask 255.255.255.255 destination 62.156.z.k
> inet6 fe80::9565:3024:4deb:ebc7 prefixlen 10 scopeid 0x20
> ppp txqueuelen 3 (Point-to-Point Protocol)
> RX packets 852397 bytes 480197087 (457.9 MiB)
> RX errors 0 dropped 0 overruns 0 frame 0
> TX packets 967912 bytes 170822532 (162.9 MiB)
> TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0
>
>> The LTE phones could have lower MTU and thus overcome PMTU problem.
>
> Should I reduce the MTU?!?
> Maybe I didn't understood what you mean...
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
> --
> _
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>
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> https://community.asterisk.org/
>
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> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] Voice "broken" during calls
Missing packet from DT could be caused by MTU issue. Marek 2020-06-18 5:41 GMT+02:00, Jeff LaCoursiere : > Hello Luca, > > We are still playing with visualization of your data, but I didn't want > you to wait any longer for some results. I think I blame both DT and > the Pi :) > > First, a look at the phone side of your Banana Pi. The first thing we > noticed is there were a LOT more packets in one direction (north towards > DT) than the other (towards the phone): > > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr > testPhone.pcap src 192.168.200.10 | wc -l > reading from file testPhone.pcap, link-type EN10MB (Ethernet) > 7951 > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr > testPhone.pcap dst 192.168.200.10 | wc -l > reading from file testPhone.pcap, link-type EN10MB (Ethernet) > 3981 > > > Note there are almost twice as many packets headed out. Our tool takes > a shot at it: > > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ wotinder -I > testPhone.pcap > input: testPhone.pcap > 2020/06/16 10:47:16.047401 INVITE 192.168.200.10:25572 (Luca) -> > 192.168.200.1:25572 (sip:035014649215)-(81b17560-c0a80101-0-1798,1) > 2020/06/16 10:47:16.112866 DUPINVITE 192.168.200.10:25572 (Luca) -> > 192.168.200.1:25572 (sip:035014649215)-(81b17560-c0a80101-0-1798,1) > 2020/06/16 10:48:43.690647 BYE 192.168.200.1:25572(sip:035014649215) > -> 192.168.200.10:25572(Luca) > Session: 81b17560-c0a80101-0-1798 > RTP 1 -> 10030 > Source total pkts: 7899 (avg err 15934.774414) > Dest total pkts: 3943 (avg err 8307.511719) > > The "average error" is the average departure from exactly 50hz, in > microseconds. Basically we are wanting to see a packet every 20,000us, > and if it arrives early (because the last one was late) or late, then > the absolute value of how far off it was is accumulated, and in the end > averaged. Its a bit misleading in this case, because there has clearly > been packet loss in one direction, and I am still wrapping my head > around why the error isn't much higher (some kind of bug in our packet > loss penalties). > > It does show that from the BPi's perspective, the stream from the phone > is NOT very steady. The *average* error was almost a full packet length > late (16,000us). Now our tool spits out the raw data (time between > packets in us) and we can quickly graph it. I lined up the two legs, > but of course you are only seeing half of the second one, and it makes > an interesting visual: > > > What on earth is causing the very regular spikes? Roughly every second > there seems to be a delay introduced, EVEN FROM THE PHONE ON THE LAN! > This worries me that we have asked the Pi to do too much. Perhaps > capturing the data and writing it while also running asterisk is causing > something to back up regularly. We do prefer to do this kind of > analysis from a span port on a switch... > > But that doesn't explain the missing packets from DT. > > Similar results on that side: > > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr > testDSL.pcap src 91.49.58.181 | wc -l > reading from file testDSL.pcap, link-type LINUX_SLL (Linux cooked) > 8048 > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ tcpdump -nr > testDSL.pcap dst 91.49.58.181 | wc -l > reading from file testDSL.pcap, link-type LINUX_SLL (Linux cooked) > 4076 > > > I'm making an assumption that 91.49.58.181 is your side of the DSL, and > the packets towards you seem to be missing a lot! I can't explain that > as a Pi issue *unless* something funny is happening on the kernel > handling inbound public traffic. You mention you are traffic shaping - > that could easily be causing something like this. Running our tool on > that trace: > > jeff@jasper:~/Personal/StratusTalk/wotinder/Luca/tmp$ wotinder -I > DSL.pcap > input: DSL.pcap > 2020/06/16 10:47:16.196746 INVITE 91.49.58.181:25572 > (00493514977290) -> 217.0.27.53:5060 > (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) > 2020/06/16 10:47:16.296309 DUPINVITE 91.49.58.181:25572 > (00493514977290) -> 217.0.27.53:5060 > (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) > 2020/06/16 10:47:16.357971 DUPINVITE 91.49.58.181:25572 > (00493514977290) -> 217.0.27.53:5060 > (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) > 2020/06/16 10:47:16.457280 DUPINVITE 91.49.58.181:25572 > (00493514977290) -> 217.0.27.53:5060 > (sip:035014649215)-(765cb6164b1c122a3b9c8303600ea367,10036) > 2020/06/16 10:48:43.680671 BYE 217.0.27.53:5060(sip:035014649215) -> > 91.49.58.181:25572(00493514977290) > Session: 765cb6164b1c122a3b9c8303600ea367 > RTP 10036 -> 6300 > Source total pkts: 7898 (avg err 15771.558594) > Dest total pkts: 3943 (avg err 6995.069824) > > > The
Re: [asterisk-users] Voice broken during calls (again...)
Would you mind repeating the test with canreinvite=no set for all you
phones and mobile phones?
What is your upload bitrate? Is it guaranteed?
I would try also to test the PMTU:
Try:
ping -M do -s 2000 ${ip address of the sip server}
You should receive icmp asking for lowering the packet size.
The LTE phones could have lower MTU and thus overcome PMTU problem.
Marek
2020-06-22 21:48 GMT+02:00, Luca Bertoncello :
> A thing I forgot to report...
> My Asterisk listen on an high port (*not* 5060), since I had many
> problems in the past with someone trying to use my Asterisk with brute
> force attack...
>
> I really don't think, this can be the problem, but better to report all...
>
> Regards
> 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
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Voice broken during calls (again...)
Hello,
try pinging your sip peer ip address following way:
ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress}
Post several lines and the statistics.
Were you also thinking about MTU problems? Not very probable, but one
never knows.
Marek
2020-06-22 17:18 GMT+02:00, Luca Bertoncello :
> Am 22.06.2020 um 17:01 schrieb Telium Technical Support:
>> I don't know if there was a prior email with more details, but
>>
>> Latency is as important as speed. Have you checked latency between your
>> device and pop? What about QoS at your location, and does your ITSP
>> support/respect QoS?
>
> That's a very good idea...
> Could you suggest me how can I check it?
> The Gateway is a Linux with Debian 9.
>
>> Could problem be inside your network? Have you tested/optimized internal?
>
> Really difficult to believe... If I call another VoIP-phone in my
> network (using the "internal number") the quality is excellent.
>
> If I call my wife using the "external number", the quality is very bad...
>
> 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
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Voice "broken" during calls
Hi Luca, I suspect the problem is either the line quality, aggregation or some other factor. I can see you allow alaw and ulaw codecs for DT and alaw, ulaw and gsm for the second provider. This could be the difference why you observe problems mainly on DT. The alaw and ulaw codec require 64 kbps stream, but gsm requires only 13 kbps. If this is true, your problems will most probably be gone right after switching to the business contract. So happy tomorow. Marek 2020-06-17 15:07 GMT+02:00, Luca Bertoncello : > Am 17.06.2020 14:37, schrieb Karsten Wemheuer: > > Hi Karsten! > >> The product is "All-IP" and not the SIP trunk, right? >> The call starts normally and after about 15 minutes the quality is >> disturbed? > > No, current we have Magenta Zuhause. Tomorrow we'll change to > DeutschlandLAN IP (business contract). > The quality is disturbed from the first second... > > I had the problem, that the connection will be *dropped* after 15 > minutes, and I solved it with "session-timers = refuse" > > Bye > 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 -- _ -- 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] call replicating
Hello, provider responded the behavior is intentional from their side. So this should be fixed in asterisk. The pjsip cleanly does not do any unregistrations where it should. Marek 2020-06-07 12:30 GMT+02:00, Marek Greško : > Hello, > > I found the problem and also the workaround. > > Clearly, since it was working with chan_sip it should not be dialplan > problem, but sip stack problem. > > I have line=yes set up. After asterisk restart the old registration is > not unregistered and new one is registered with different line value. > Then incoming invites and qualify requests are sent to all the > registrations and there the problem lies. > > I am thinking of how could asterisk prevent such situations. > > 1. I think it should send unregistration requests on shutdown. > > 2. I think it should keep the database of active registrations and > unregister and reregister all of them during startup in case some of > them remain active after unclean shutdown. > > Also probably provider side should be fixed? > > Thanks for your insight. > > Marek > > > 2020-06-05 19:29 GMT+02:00, Doug Lytle : >> On 6/5/20 12:24 PM, Marek Greško wrote: >>> How can this behavior been overriden? I do not expect this is problem >>> on provider side, since it was working normally using chan_sip. >> >> Console output and dial plan snippets are always useful when diagnosing, >> >> Doug >> >> > -- _ -- 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] call replicating
Hello, I found the problem and also the workaround. Clearly, since it was working with chan_sip it should not be dialplan problem, but sip stack problem. I have line=yes set up. After asterisk restart the old registration is not unregistered and new one is registered with different line value. Then incoming invites and qualify requests are sent to all the registrations and there the problem lies. I am thinking of how could asterisk prevent such situations. 1. I think it should send unregistration requests on shutdown. 2. I think it should keep the database of active registrations and unregister and reregister all of them during startup in case some of them remain active after unclean shutdown. Also probably provider side should be fixed? Thanks for your insight. Marek 2020-06-05 19:29 GMT+02:00, Doug Lytle : > On 6/5/20 12:24 PM, Marek Greško wrote: >> How can this behavior been overriden? I do not expect this is problem >> on provider side, since it was working normally using chan_sip. > > Console output and dial plan snippets are always useful when diagnosing, > > Doug > > -- _ -- 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 replicating
Hello, after migration from chan_sip to res_pjsip I get strange behavior when receiving call from the outside world. When call is received, it is replicated multiple times. Two of that calls get to the phone. So the phone is ringing on both lines. When having only Dial function in dialplan I am able to place call. But when creating some dialplan procedures containing VoiceMail I get phone ringing for 1 second and it stops. The caller is immediatelly directed to voicemail. It is because the second (or third) call gets busy. How can this behavior been overriden? I do not expect this is problem on provider side, since it was working normally using chan_sip. Thanks Marek -- _ -- 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] pjsip subscribecontext support
Hello, great news. I did not find it because an underscore added compared to chan_sip. Thank you very much. It is working. Marek 2020-06-05 11:12 GMT+02:00, Joshua C. Colp : > On Fri, Jun 5, 2020 at 6:02 AM Marek Greško wrote: > >> Hello, >> >> I would like to ask about current state of subscribecontext in pjsip. >> I found out some 6 years old discussion on that without any plans to >> implement it in the future. >> >> I have phones in different contexts. I suspect, when I use its context >> to subscribe, they will not see phones from the different contexts. Am >> I right? >> > > I don't recall when the option was implemented but it's present on the > endpoint[1]. > > [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip#Asterisk16Configuration_res_pjsip-endpoint_subscribe_context > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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] pjsip subscribecontext support
Hello, I would like to ask about current state of subscribecontext in pjsip. I found out some 6 years old discussion on that without any plans to implement it in the future. I have phones in different contexts. I suspect, when I use its context to subscribe, they will not see phones from the different contexts. Am I right? Marek -- _ -- 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
