Re: [OpenSIPS-Users] xlog message sent to homer 7 via HEP has type as ERROR
Hello Liviu Chircu and Team,
I have recompiled and checked with said commit of opensips
opensips -V
version: opensips 3.1.1 (x86_64/linux)
git revision: 2212865f1
main.c compiled on 07:18:08 Jan 15 2021 with gcc 7
Now the logs are seen as NOTICE and not ERROR.
L_INFO does not make it to HEP logs
L_ALERT makes it to HEP as NOTICE
and other logs used as xlog("Anything ") also make it as NOTICE.
should they look like this or L_INFO should be seen as INFO while L_ALERT
seen as ALERT .
Sunil More
Manager - DevOps
91 95033 38275
sunil.m...@samespace.com
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Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Thanks for the responses. They helped me exclude some things. I've managed to make progress and pinned down the lack of audio to a misconfiguration of Mediaproxy. Two-way audio through double-nat / firewall is working but goes silent after about 60 seconds connected and Asterisk kills the connection 31 seconds later due to lack of RTP activity for the last 31 seconds On Thu, 14 Jan 2021 at 12:00, David Villasmil < david.villasmil.w...@gmail.com> wrote: > Check out what IPs are offered in the SDPs in asterisk. Make sure they’re > both public IPs. > If you only have 1 asterisk, forwarding the rtp port range configured in > asterisk from the firewall to asterisk should do it. > > > On Thu, 14 Jan 2021 at 08:23, Mark Allen wrote: > >> Thanks Adrian >> >> The firewall has SIP-ALG disabled and just forwards ports from externally >> to where they need to be internally - so ports 5060 and 1 - 65535 of >> 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box) >> >> On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu >> wrote: >> >>> Google search for SIP ALG problem to see if this is relevant for your >>> case. >>> >>> Regards, >>> Adrian >>> >>> >>> On 13 Jan 2021, at 13:08, Mark Allen wrote: >>> >>> Hi all - I've been banging my head against this but not succeeding. >>> >>> Our setup... >>> >>> UAC 192.168.x.x >>> | >>> Router5.x.x.x >>> | >>> (internet) >>> | >>> Firewall 46.x.x.x maps >>> | directly to >>> OpenSIPS 192.168.x.x Mid-registrar >>> | >>> Asterisk 192.168.x.x >>> >>> >>> Current situation: >>> - UAC can register on Asterisk via OpenSIPS >>> - UAC can call destination registered on Asterisk on local n/w to >>> Asterisk box >>> - Destination extension rings and can pick up call >>> - There is no audio either way & call drops after about 30 secs >>> (Asterisk kills call with "Requested channel not available" because not >>> RTP traffic is reaching destination) >>> >>> I have tried passing audio through Mediaproxy on OpenSIPS box but with >>> no success. Using Wireshark I can see RTP traffic initiated at both ends, >>> but it doesn't reach the other end either way. >>> >>> Is there some definitive guide to setting this up correctly or are there >>> specific steps that I need to follow? >>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -- > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Check out what IPs are offered in the SDPs in asterisk. Make sure they’re both public IPs. If you only have 1 asterisk, forwarding the rtp port range configured in asterisk from the firewall to asterisk should do it. On Thu, 14 Jan 2021 at 08:23, Mark Allen wrote: > Thanks Adrian > > The firewall has SIP-ALG disabled and just forwards ports from externally > to where they need to be internally - so ports 5060 and 1 - 65535 of > 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box) > > On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu wrote: > >> Google search for SIP ALG problem to see if this is relevant for your >> case. >> >> Regards, >> Adrian >> >> >> On 13 Jan 2021, at 13:08, Mark Allen wrote: >> >> Hi all - I've been banging my head against this but not succeeding. >> >> Our setup... >> >> UAC 192.168.x.x >> | >> Router5.x.x.x >> | >> (internet) >> | >> Firewall 46.x.x.x maps >> | directly to >> OpenSIPS 192.168.x.x Mid-registrar >> | >> Asterisk 192.168.x.x >> >> >> Current situation: >> - UAC can register on Asterisk via OpenSIPS >> - UAC can call destination registered on Asterisk on local n/w to >> Asterisk box >> - Destination extension rings and can pick up call >> - There is no audio either way & call drops after about 30 secs (Asterisk >> kills call with "Requested channel not available" because not RTP >> traffic is reaching destination) >> >> I have tried passing audio through Mediaproxy on OpenSIPS box but with no >> success. Using Wireshark I can see RTP traffic initiated at both ends, but >> it doesn't reach the other end either way. >> >> Is there some definitive guide to setting this up correctly or are there >> specific steps that I need to follow? >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Thanks Adrian The firewall has SIP-ALG disabled and just forwards ports from externally to where they need to be internally - so ports 5060 and 1 - 65535 of 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box) On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu wrote: > Google search for SIP ALG problem to see if this is relevant for your case. > > Regards, > Adrian > > > On 13 Jan 2021, at 13:08, Mark Allen wrote: > > Hi all - I've been banging my head against this but not succeeding. > > Our setup... > > UAC 192.168.x.x > | > Router5.x.x.x > | > (internet) > | > Firewall 46.x.x.x maps > | directly to > OpenSIPS 192.168.x.x Mid-registrar > | > Asterisk 192.168.x.x > > > Current situation: > - UAC can register on Asterisk via OpenSIPS > - UAC can call destination registered on Asterisk on local n/w to Asterisk > box > - Destination extension rings and can pick up call > - There is no audio either way & call drops after about 30 secs (Asterisk > kills call with "Requested channel not available" because not RTP traffic > is reaching destination) > > I have tried passing audio through Mediaproxy on OpenSIPS box but with no > success. Using Wireshark I can see RTP traffic initiated at both ends, but > it doesn't reach the other end either way. > > Is there some definitive guide to setting this up correctly or are there > specific steps that I need to follow? > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
