Re: [asterisk-users] asterisk server - no sound
And it is worst (and that could be the reason of the error). 127.0.0.1 is configured in 2 interfaces (lo and venet0), just with different network masks. Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 18:54, andre castro wrote: > I am using version: 14.5.0 > No, Im not using Dundi. > Can you a bit more informative when you say I "need to configure the IPs > in your server"? > thanks! > a > On 06/06/2017 07:47 PM, Marcelo Terres wrote: >> I think you need to configure the IPs in your server. You just have >> localhost... >> Marcelo H. Terres >> IM: mhter...@jabber.mundoopensource.com.br >> https://www.mundoopensource.com.br >> https://twitter.com/mhterres >> https://linkedin.com/in/marceloterres >> >> >> On 6 June 2017 at 16:27, andre castro wrote: >>> Thanks Anthony. >>> >>> I did it on the server, according to >>> https://www.voip-info.org/wiki/view/port+forwarding >>> >>> However after doing it, when running Asterisk I get the following message >>> sudo asterisk -vvr >>> No ethernet interface found for seeding global EID. You will have to set >>> it manually. >>> Unable to access the running directory (No such file or directory). >>> Changing to '/' for compatibility. >>> >>> How and where can it be set? >>> >>> My server ifconfig: >>> >>> loLink encap:Local Loopback >>> inet addr:127.0.0.1 Mask:255.0.0.0 >>> inet6 addr: ::1/128 Scope:Host >>> UP LOOPBACK RUNNING MTU:65536 Metric:1 >>> RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 >>> TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 >>> collisions:0 txqueuelen:0 >>> RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) >>> >>> venet0Link encap:UNSPEC HWaddr >>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >>> inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 >>> Mask:255.255.255.255 >>> inet6 addr: ::2/128 Scope:Compat >>> inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global >>> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >>> RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 >>> TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 >>> collisions:0 txqueuelen:0 >>> RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) >>> >>> venet0:0 Link encap:UNSPEC HWaddr >>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >>> inet addr:server.ip.add.r P-t-P:server.ip.add.r >>> Bcast:server.ip.add.r Mask:255.255.255.255 >>> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >>> >>> >>> >>> On 06/06/2017 05:09 PM, Antony Stone wrote: On Tuesday 06 June 2017 16:57:07 andre castro wrote: > On 06/06/2017 04:36 PM, Antony Stone wrote: >> >> Tell us about your networking arrangement - are both phones and the >> Asterisk machine on the same network? > > Nop. They are in 2 different networks. The phones in one and the > Asterisk machine in another. Okay, that is why you have audio between the two phones, then - they can see each other directly, on the same network, and nothing is interfering with the traffic between them. >> Is there a router in between any of them? > > Yes. In the phones network. > >> Is there any NAT involved? > > Yes in the phones' network. They both have different private IP address > and one public IP. Okay, I suspect that this NATting router is not passing the UDP packets from the server back to the phones correctly, based on the SIP connection (when the phone makes the call). SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. If it's a Linux router, you need to make sure you are allowing FORWARDed traffic which matches ESTABLISHED, RELATED. If it's not a Linux router, you need to find out how to get it to support SIP and RTSP. Good luck, Antony. >>> >>> -- >>> oo.io >>> bibliotecha.info >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Check out the new Asterisk community forum at: >>> https://community.asterisk.org/ >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > oo.io > bibliotecha.info > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new
Re: [asterisk-users] asterisk server - no sound
Well, based on the config that you sent, your server just have the localhost IP (127.0.0.1) Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 18:54, andre castro wrote: > I am using version: 14.5.0 > No, Im not using Dundi. > Can you a bit more informative when you say I "need to configure the IPs > in your server"? > thanks! > a > On 06/06/2017 07:47 PM, Marcelo Terres wrote: >> I think you need to configure the IPs in your server. You just have >> localhost... >> Marcelo H. Terres >> IM: mhter...@jabber.mundoopensource.com.br >> https://www.mundoopensource.com.br >> https://twitter.com/mhterres >> https://linkedin.com/in/marceloterres >> >> >> On 6 June 2017 at 16:27, andre castro wrote: >>> Thanks Anthony. >>> >>> I did it on the server, according to >>> https://www.voip-info.org/wiki/view/port+forwarding >>> >>> However after doing it, when running Asterisk I get the following message >>> sudo asterisk -vvr >>> No ethernet interface found for seeding global EID. You will have to set >>> it manually. >>> Unable to access the running directory (No such file or directory). >>> Changing to '/' for compatibility. >>> >>> How and where can it be set? >>> >>> My server ifconfig: >>> >>> loLink encap:Local Loopback >>> inet addr:127.0.0.1 Mask:255.0.0.0 >>> inet6 addr: ::1/128 Scope:Host >>> UP LOOPBACK RUNNING MTU:65536 Metric:1 >>> RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 >>> TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 >>> collisions:0 txqueuelen:0 >>> RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) >>> >>> venet0Link encap:UNSPEC HWaddr >>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >>> inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 >>> Mask:255.255.255.255 >>> inet6 addr: ::2/128 Scope:Compat >>> inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global >>> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >>> RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 >>> TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 >>> collisions:0 txqueuelen:0 >>> RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) >>> >>> venet0:0 Link encap:UNSPEC HWaddr >>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >>> inet addr:server.ip.add.r P-t-P:server.ip.add.r >>> Bcast:server.ip.add.r Mask:255.255.255.255 >>> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >>> >>> >>> >>> On 06/06/2017 05:09 PM, Antony Stone wrote: On Tuesday 06 June 2017 16:57:07 andre castro wrote: > On 06/06/2017 04:36 PM, Antony Stone wrote: >> >> Tell us about your networking arrangement - are both phones and the >> Asterisk machine on the same network? > > Nop. They are in 2 different networks. The phones in one and the > Asterisk machine in another. Okay, that is why you have audio between the two phones, then - they can see each other directly, on the same network, and nothing is interfering with the traffic between them. >> Is there a router in between any of them? > > Yes. In the phones network. > >> Is there any NAT involved? > > Yes in the phones' network. They both have different private IP address > and one public IP. Okay, I suspect that this NATting router is not passing the UDP packets from the server back to the phones correctly, based on the SIP connection (when the phone makes the call). SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. If it's a Linux router, you need to make sure you are allowing FORWARDed traffic which matches ESTABLISHED, RELATED. If it's not a Linux router, you need to find out how to get it to support SIP and RTSP. Good luck, Antony. >>> >>> -- >>> oo.io >>> bibliotecha.info >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Check out the new Asterisk community forum at: >>> https://community.asterisk.org/ >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > oo.io > bibliotecha.info > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ >
Re: [asterisk-users] asterisk server - no sound
I am using version: 14.5.0 No, Im not using Dundi. Can you a bit more informative when you say I "need to configure the IPs in your server"? thanks! a On 06/06/2017 07:47 PM, Marcelo Terres wrote: > I think you need to configure the IPs in your server. You just have > localhost... > Marcelo H. Terres > IM: mhter...@jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres > https://linkedin.com/in/marceloterres > > > On 6 June 2017 at 16:27, andre castro wrote: >> Thanks Anthony. >> >> I did it on the server, according to >> https://www.voip-info.org/wiki/view/port+forwarding >> >> However after doing it, when running Asterisk I get the following message >> sudo asterisk -vvr >> No ethernet interface found for seeding global EID. You will have to set >> it manually. >> Unable to access the running directory (No such file or directory). >> Changing to '/' for compatibility. >> >> How and where can it be set? >> >> My server ifconfig: >> >> loLink encap:Local Loopback >> inet addr:127.0.0.1 Mask:255.0.0.0 >> inet6 addr: ::1/128 Scope:Host >> UP LOOPBACK RUNNING MTU:65536 Metric:1 >> RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 >> TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 >> collisions:0 txqueuelen:0 >> RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) >> >> venet0Link encap:UNSPEC HWaddr >> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >> inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 >> Mask:255.255.255.255 >> inet6 addr: ::2/128 Scope:Compat >> inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global >> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >> RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 >> TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 >> collisions:0 txqueuelen:0 >> RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) >> >> venet0:0 Link encap:UNSPEC HWaddr >> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >> inet addr:server.ip.add.r P-t-P:server.ip.add.r >> Bcast:server.ip.add.r Mask:255.255.255.255 >> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >> >> >> >> On 06/06/2017 05:09 PM, Antony Stone wrote: >>> On Tuesday 06 June 2017 16:57:07 andre castro wrote: >>> On 06/06/2017 04:36 PM, Antony Stone wrote: > > Tell us about your networking arrangement - are both phones and the > Asterisk machine on the same network? Nop. They are in 2 different networks. The phones in one and the Asterisk machine in another. >>> >>> Okay, that is why you have audio between the two phones, then - they can see >>> each other directly, on the same network, and nothing is interfering with >>> the >>> traffic between them. >>> > Is there a router in between any of them? Yes. In the phones network. > Is there any NAT involved? Yes in the phones' network. They both have different private IP address and one public IP. >>> >>> Okay, I suspect that this NATting router is not passing the UDP packets from >>> the server back to the phones correctly, based on the SIP connection (when >>> the >>> phone makes the call). >>> >>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. >>> >>> If it's a Linux router, you need to make sure you are allowing FORWARDed >>> traffic >>> which matches ESTABLISHED, RELATED. >>> >>> If it's not a Linux router, you need to find out how to get it to support >>> SIP >>> and RTSP. >>> >>> >>> Good luck, >>> >>> >>> Antony. >>> >> >> -- >> oo.io >> bibliotecha.info >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > -- oo.io bibliotecha.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk server - no sound
I think you need to configure the IPs in your server. You just have localhost... Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 16:27, andre castro wrote: > Thanks Anthony. > > I did it on the server, according to > https://www.voip-info.org/wiki/view/port+forwarding > > However after doing it, when running Asterisk I get the following message > sudo asterisk -vvr > No ethernet interface found for seeding global EID. You will have to set > it manually. > Unable to access the running directory (No such file or directory). > Changing to '/' for compatibility. > > How and where can it be set? > > My server ifconfig: > > loLink encap:Local Loopback > inet addr:127.0.0.1 Mask:255.0.0.0 > inet6 addr: ::1/128 Scope:Host > UP LOOPBACK RUNNING MTU:65536 Metric:1 > RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 > TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 > collisions:0 txqueuelen:0 > RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) > > venet0Link encap:UNSPEC HWaddr > 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 > inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 > Mask:255.255.255.255 > inet6 addr: ::2/128 Scope:Compat > inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global > UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 > RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 > TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 > collisions:0 txqueuelen:0 > RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) > > venet0:0 Link encap:UNSPEC HWaddr > 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 > inet addr:server.ip.add.r P-t-P:server.ip.add.r > Bcast:server.ip.add.r Mask:255.255.255.255 > UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 > > > > On 06/06/2017 05:09 PM, Antony Stone wrote: >> On Tuesday 06 June 2017 16:57:07 andre castro wrote: >> >>> On 06/06/2017 04:36 PM, Antony Stone wrote: Tell us about your networking arrangement - are both phones and the Asterisk machine on the same network? >>> >>> Nop. They are in 2 different networks. The phones in one and the >>> Asterisk machine in another. >> >> Okay, that is why you have audio between the two phones, then - they can see >> each other directly, on the same network, and nothing is interfering with the >> traffic between them. >> Is there a router in between any of them? >>> >>> Yes. In the phones network. >>> Is there any NAT involved? >>> >>> Yes in the phones' network. They both have different private IP address >>> and one public IP. >> >> Okay, I suspect that this NATting router is not passing the UDP packets from >> the server back to the phones correctly, based on the SIP connection (when >> the >> phone makes the call). >> >> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. >> >> If it's a Linux router, you need to make sure you are allowing FORWARDed >> traffic >> which matches ESTABLISHED, RELATED. >> >> If it's not a Linux router, you need to find out how to get it to support SIP >> and RTSP. >> >> >> Good luck, >> >> >> Antony. >> > > -- > oo.io > bibliotecha.info > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 server - no sound
Looks like it comes com pbx_dundi.c. Why are you using dundi? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 18:43, Marcelo Terres wrote: > Which Asterisk version are you using? > > Marcelo H. Terres > IM: mhter...@jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres > https://linkedin.com/in/marceloterres > > > On 6 June 2017 at 18:32, andre castro wrote: >> Any ideas. >> After configuring port forwarding on the server (machine making nat) to >> forward connections originated from external clients to the machine >> running asterisk, as explained in >> https://www.voip-info.org/wiki/view/port+forwarding >> My peers were unable to register. >> >> >> And When running Asterisk I am getting: >> No ethernet interface found for seeding global EID. You will have to set >> it manually. >> Unable to access the running directory (No such file or directory). >> Changing to '/' for compatibility. >> >> Any advice what to do next? >> >> thanks >> a >> >> On 06/06/2017 05:27 PM, andre castro wrote: >>> Thanks Anthony. >>> >>> I did it on the server, according to >>> https://www.voip-info.org/wiki/view/port+forwarding >>> >>> However after doing it, when running Asterisk I get the following message >>> sudo asterisk -vvr >>> No ethernet interface found for seeding global EID. You will have to set >>> it manually. >>> Unable to access the running directory (No such file or directory). >>> Changing to '/' for compatibility. >>> >>> How and where can it be set? >>> >>> My server ifconfig: >>> >>> loLink encap:Local Loopback >>> inet addr:127.0.0.1 Mask:255.0.0.0 >>> inet6 addr: ::1/128 Scope:Host >>> UP LOOPBACK RUNNING MTU:65536 Metric:1 >>> RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 >>> TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 >>> collisions:0 txqueuelen:0 >>> RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) >>> >>> venet0Link encap:UNSPEC HWaddr >>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >>> inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 >>> Mask:255.255.255.255 >>> inet6 addr: ::2/128 Scope:Compat >>> inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global >>> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >>> RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 >>> TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 >>> collisions:0 txqueuelen:0 >>> RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) >>> >>> venet0:0 Link encap:UNSPEC HWaddr >>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >>> inet addr:server.ip.add.r P-t-P:server.ip.add.r >>> Bcast:server.ip.add.r Mask:255.255.255.255 >>> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >>> >>> >>> >>> On 06/06/2017 05:09 PM, Antony Stone wrote: On Tuesday 06 June 2017 16:57:07 andre castro wrote: > On 06/06/2017 04:36 PM, Antony Stone wrote: >> >> Tell us about your networking arrangement - are both phones and the >> Asterisk machine on the same network? > > Nop. They are in 2 different networks. The phones in one and the > Asterisk machine in another. Okay, that is why you have audio between the two phones, then - they can see each other directly, on the same network, and nothing is interfering with the traffic between them. >> Is there a router in between any of them? > > Yes. In the phones network. > >> Is there any NAT involved? > > Yes in the phones' network. They both have different private IP address > and one public IP. Okay, I suspect that this NATting router is not passing the UDP packets from the server back to the phones correctly, based on the SIP connection (when the phone makes the call). SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. If it's a Linux router, you need to make sure you are allowing FORWARDed traffic which matches ESTABLISHED, RELATED. If it's not a Linux router, you need to find out how to get it to support SIP and RTSP. Good luck, Antony. >>> >> >> -- >> oo.io >> bibliotecha.info >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visi
Re: [asterisk-users] asterisk server - no sound
Which Asterisk version are you using? Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 18:32, andre castro wrote: > Any ideas. > After configuring port forwarding on the server (machine making nat) to > forward connections originated from external clients to the machine > running asterisk, as explained in > https://www.voip-info.org/wiki/view/port+forwarding > My peers were unable to register. > > > And When running Asterisk I am getting: > No ethernet interface found for seeding global EID. You will have to set > it manually. > Unable to access the running directory (No such file or directory). > Changing to '/' for compatibility. > > Any advice what to do next? > > thanks > a > > On 06/06/2017 05:27 PM, andre castro wrote: >> Thanks Anthony. >> >> I did it on the server, according to >> https://www.voip-info.org/wiki/view/port+forwarding >> >> However after doing it, when running Asterisk I get the following message >> sudo asterisk -vvr >> No ethernet interface found for seeding global EID. You will have to set >> it manually. >> Unable to access the running directory (No such file or directory). >> Changing to '/' for compatibility. >> >> How and where can it be set? >> >> My server ifconfig: >> >> loLink encap:Local Loopback >> inet addr:127.0.0.1 Mask:255.0.0.0 >> inet6 addr: ::1/128 Scope:Host >> UP LOOPBACK RUNNING MTU:65536 Metric:1 >> RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 >> TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 >> collisions:0 txqueuelen:0 >> RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) >> >> venet0Link encap:UNSPEC HWaddr >> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >> inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 >> Mask:255.255.255.255 >> inet6 addr: ::2/128 Scope:Compat >> inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global >> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >> RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 >> TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 >> collisions:0 txqueuelen:0 >> RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) >> >> venet0:0 Link encap:UNSPEC HWaddr >> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 >> inet addr:server.ip.add.r P-t-P:server.ip.add.r >> Bcast:server.ip.add.r Mask:255.255.255.255 >> UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 >> >> >> >> On 06/06/2017 05:09 PM, Antony Stone wrote: >>> On Tuesday 06 June 2017 16:57:07 andre castro wrote: >>> On 06/06/2017 04:36 PM, Antony Stone wrote: > > Tell us about your networking arrangement - are both phones and the > Asterisk machine on the same network? Nop. They are in 2 different networks. The phones in one and the Asterisk machine in another. >>> >>> Okay, that is why you have audio between the two phones, then - they can see >>> each other directly, on the same network, and nothing is interfering with >>> the >>> traffic between them. >>> > Is there a router in between any of them? Yes. In the phones network. > Is there any NAT involved? Yes in the phones' network. They both have different private IP address and one public IP. >>> >>> Okay, I suspect that this NATting router is not passing the UDP packets from >>> the server back to the phones correctly, based on the SIP connection (when >>> the >>> phone makes the call). >>> >>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. >>> >>> If it's a Linux router, you need to make sure you are allowing FORWARDed >>> traffic >>> which matches ESTABLISHED, RELATED. >>> >>> If it's not a Linux router, you need to find out how to get it to support >>> SIP >>> and RTSP. >>> >>> >>> Good luck, >>> >>> >>> Antony. >>> >> > > -- > oo.io > bibliotecha.info > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 vi
Re: [asterisk-users] asterisk server - no sound
Any ideas. After configuring port forwarding on the server (machine making nat) to forward connections originated from external clients to the machine running asterisk, as explained in https://www.voip-info.org/wiki/view/port+forwarding My peers were unable to register. And When running Asterisk I am getting: No ethernet interface found for seeding global EID. You will have to set it manually. Unable to access the running directory (No such file or directory). Changing to '/' for compatibility. Any advice what to do next? thanks a On 06/06/2017 05:27 PM, andre castro wrote: > Thanks Anthony. > > I did it on the server, according to > https://www.voip-info.org/wiki/view/port+forwarding > > However after doing it, when running Asterisk I get the following message > sudo asterisk -vvr > No ethernet interface found for seeding global EID. You will have to set > it manually. > Unable to access the running directory (No such file or directory). > Changing to '/' for compatibility. > > How and where can it be set? > > My server ifconfig: > > loLink encap:Local Loopback > inet addr:127.0.0.1 Mask:255.0.0.0 > inet6 addr: ::1/128 Scope:Host > UP LOOPBACK RUNNING MTU:65536 Metric:1 > RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 > TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 > collisions:0 txqueuelen:0 > RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) > > venet0Link encap:UNSPEC HWaddr > 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 > inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 > Mask:255.255.255.255 > inet6 addr: ::2/128 Scope:Compat > inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global > UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 > RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 > TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 > collisions:0 txqueuelen:0 > RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) > > venet0:0 Link encap:UNSPEC HWaddr > 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 > inet addr:server.ip.add.r P-t-P:server.ip.add.r > Bcast:server.ip.add.r Mask:255.255.255.255 > UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 > > > > On 06/06/2017 05:09 PM, Antony Stone wrote: >> On Tuesday 06 June 2017 16:57:07 andre castro wrote: >> >>> On 06/06/2017 04:36 PM, Antony Stone wrote: Tell us about your networking arrangement - are both phones and the Asterisk machine on the same network? >>> >>> Nop. They are in 2 different networks. The phones in one and the >>> Asterisk machine in another. >> >> Okay, that is why you have audio between the two phones, then - they can see >> each other directly, on the same network, and nothing is interfering with >> the >> traffic between them. >> Is there a router in between any of them? >>> >>> Yes. In the phones network. >>> Is there any NAT involved? >>> >>> Yes in the phones' network. They both have different private IP address >>> and one public IP. >> >> Okay, I suspect that this NATting router is not passing the UDP packets from >> the server back to the phones correctly, based on the SIP connection (when >> the >> phone makes the call). >> >> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. >> >> If it's a Linux router, you need to make sure you are allowing FORWARDed >> traffic >> which matches ESTABLISHED, RELATED. >> >> If it's not a Linux router, you need to find out how to get it to support >> SIP >> and RTSP. >> >> >> Good luck, >> >> >> Antony. >> > -- oo.io bibliotecha.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk server - no sound
Try to use the echo app. If you can listen your echo, so it is something in the network. Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 14:18, andre castro wrote: > hello folks, > this might be a simple question... > > I just installed asterisk in a debian server. > All seems to be running fine, but the audio sent by the server. > If I have one of my registered peers call and extension (102) that plays > back audio (extension.conf and sip.conf coffee-pasted below), Asterisk > answers and prints no errors. > Its `sip show channels` prints: > > PeerUser/ANRCall IDFormatHoldLast MessageExpiry >Peer > peer.ip1001 1...-5060 (ulaw) No Rx: ACK >1001 > > But I hear nothing at the peer's end. > > When one peer calls another, sound comes through just fine. > So my hunch is that is something to do with the audio supplied by the > server. > Do I need to have alsa installed?? > Any hint? > > sip.conf: > > [general] > context = unauthenticated > bindport = 5060 > bindaddr = 0.0.0.0 > tcpbindaddr = 0.0.0.0 > tcpenable = yes > videosupport = no > textsupport=yes > alwaysauthreject=yes > allowguest=no > > [1001] ; grandstream 1 > context = home > type = friend > callerid = One <1001> > secret = XYZ > host = dynamic > mailbox = 1001 > disallow = all > allow = ulaw > transport = udp > dtmfmode=auto ; accept touch-tones from the devices, negotiated > automatically > nat=force_rport > > [1005] ; mobile > context = home > type = friend > callerid = Five <1005> > secret = XYZ > host = dynamic > mailbox = 1005 > disallow = all > allow = ulaw > transport = udp > dtmfmode=auto ; accept touch-tones from the devices, negotiated > automatically > nat=force_rport > > > extensions.conf: > [home] > exten = 102,1,Answer() > same = n,Wait(1) > same = n,Playback(beep) > same = n,Wait(1) > same = n,Playback(im-sorry) > same = n,Wait(1) > same = n,Playback(number-not-answering) > same = n,Wait(1) > same = n,Hangup() > > exten => 1001,1,Dial(SIP/1001) ; grandstream phone > exten => 1005,1,Dial(SIP/1005) ; mobile > > > > > -- > oo.io > bibliotecha.info > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 server - no sound
Thanks Anthony. I did it on the server, according to https://www.voip-info.org/wiki/view/port+forwarding However after doing it, when running Asterisk I get the following message sudo asterisk -vvr No ethernet interface found for seeding global EID. You will have to set it manually. Unable to access the running directory (No such file or directory). Changing to '/' for compatibility. How and where can it be set? My server ifconfig: loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:65536 Metric:1 RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0 TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:36041459269 (33.5 GiB) TX bytes:36041459269 (33.5 GiB) venet0Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 Mask:255.255.255.255 inet6 addr: ::2/128 Scope:Compat inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0 TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:61233254724 (57.0 GiB) TX bytes:106403959440 (99.0 GiB) venet0:0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:server.ip.add.r P-t-P:server.ip.add.r Bcast:server.ip.add.r Mask:255.255.255.255 UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 On 06/06/2017 05:09 PM, Antony Stone wrote: > On Tuesday 06 June 2017 16:57:07 andre castro wrote: > >> On 06/06/2017 04:36 PM, Antony Stone wrote: >>> >>> Tell us about your networking arrangement - are both phones and the >>> Asterisk machine on the same network? >> >> Nop. They are in 2 different networks. The phones in one and the >> Asterisk machine in another. > > Okay, that is why you have audio between the two phones, then - they can see > each other directly, on the same network, and nothing is interfering with the > traffic between them. > >>> Is there a router in between any of them? >> >> Yes. In the phones network. >> >>> Is there any NAT involved? >> >> Yes in the phones' network. They both have different private IP address >> and one public IP. > > Okay, I suspect that this NATting router is not passing the UDP packets from > the server back to the phones correctly, based on the SIP connection (when > the > phone makes the call). > > SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. > > If it's a Linux router, you need to make sure you are allowing FORWARDed > traffic > which matches ESTABLISHED, RELATED. > > If it's not a Linux router, you need to find out how to get it to support SIP > and RTSP. > > > Good luck, > > > Antony. > -- oo.io bibliotecha.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk server - no sound
On Tuesday 06 June 2017 16:57:07 andre castro wrote: > On 06/06/2017 04:36 PM, Antony Stone wrote: > > > > Tell us about your networking arrangement - are both phones and the > > Asterisk machine on the same network? > > Nop. They are in 2 different networks. The phones in one and the > Asterisk machine in another. Okay, that is why you have audio between the two phones, then - they can see each other directly, on the same network, and nothing is interfering with the traffic between them. > > Is there a router in between any of them? > > Yes. In the phones network. > > > Is there any NAT involved? > > Yes in the phones' network. They both have different private IP address > and one public IP. Okay, I suspect that this NATting router is not passing the UDP packets from the server back to the phones correctly, based on the SIP connection (when the phone makes the call). SIP is on UDP 5060; audio is on UDP 10,000 - 20,000. If it's a Linux router, you need to make sure you are allowing FORWARDed traffic which matches ESTABLISHED, RELATED. If it's not a Linux router, you need to find out how to get it to support SIP and RTSP. Good luck, Antony. -- There's a good theatrical performance about puns on in the West End. It's a play on words. 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] asterisk server - no sound
Thank you Daniel for pointing out the errors and debug option. Both fixed and on. It made no difference. There are no errors printed and still no sound on ppers Now to Antony questions: On 06/06/2017 04:36 PM, Antony Stone wrote: > On Tuesday 06 June 2017 15:18:32 andre castro wrote: > >> I just installed asterisk in a debian server. >> All seems to be running fine, but the audio sent by the server. > >> But I hear nothing at the peer's end. >> >> When one peer calls another, sound comes through just fine. > > Tell us about your networking arrangement - are both phones and the Asterisk > machine on the same network? Nop. They are in 2 different networks. The phones in one and the Asterisk machine in another. > > Is there a router in between any of them? Yes. In the phones network. > > Is there any NAT involved? Yes in the phones' network. They both have different private IP address and one public IP. > >> Do I need to have alsa installed?? > > No. So I thought. Thanks guys!! > > > Antony. > -- oo.io bibliotecha.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk server - no sound
Le 06/06/2017 à 16:25, Daniel Tryba a écrit : On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote: extensions.conf: [home] exten = 102,1,Answer() same = n,Wait(1) If this is copy and paste, then your dialplan is broken (= should be =>) Well, no. = or => are the same. -- 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
Re: [asterisk-users] asterisk server - no sound
On Tuesday 06 June 2017 15:18:32 andre castro wrote: > I just installed asterisk in a debian server. > All seems to be running fine, but the audio sent by the server. > But I hear nothing at the peer's end. > > When one peer calls another, sound comes through just fine. Tell us about your networking arrangement - are both phones and the Asterisk machine on the same network? Is there a router in between any of them? Is there any NAT involved? > Do I need to have alsa installed?? No. 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
Re: [asterisk-users] asterisk server - no sound
On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote: > extensions.conf: > [home] > exten = 102,1,Answer() > same = n,Wait(1) If this is copy and paste, then your dialplan is broken (= should be =>) But to debug, enable logging (core set verbose 5), when needed debugging (core set debug 5) and sip logging (sip set debug on / pjsip set logger on). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk server - no sound
hello folks, this might be a simple question... I just installed asterisk in a debian server. All seems to be running fine, but the audio sent by the server. If I have one of my registered peers call and extension (102) that plays back audio (extension.conf and sip.conf coffee-pasted below), Asterisk answers and prints no errors. Its `sip show channels` prints: PeerUser/ANRCall IDFormatHoldLast MessageExpiry Peer peer.ip1001 1...-5060 (ulaw) No Rx: ACK 1001 But I hear nothing at the peer's end. When one peer calls another, sound comes through just fine. So my hunch is that is something to do with the audio supplied by the server. Do I need to have alsa installed?? Any hint? sip.conf: [general] context = unauthenticated bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes videosupport = no textsupport=yes alwaysauthreject=yes allowguest=no [1001] ; grandstream 1 context = home type = friend callerid = One <1001> secret = XYZ host = dynamic mailbox = 1001 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport [1005] ; mobile context = home type = friend callerid = Five <1005> secret = XYZ host = dynamic mailbox = 1005 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport extensions.conf: [home] exten = 102,1,Answer() same = n,Wait(1) same = n,Playback(beep) same = n,Wait(1) same = n,Playback(im-sorry) same = n,Wait(1) same = n,Playback(number-not-answering) same = n,Wait(1) same = n,Hangup() exten => 1001,1,Dial(SIP/1001) ; grandstream phone exten => 1005,1,Dial(SIP/1005) ; mobile -- oo.io bibliotecha.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users