# Asterisk within docker **URL:** \ **Category:** Asterisk SIP\ **Created:** [February 4, 2018, 3:35pm UTC](https://community.asterisk.org/t/asterisk-within-docker/73412 "2018-02-04T15:35:34Z")\ **Posts on this page:** 10\ **Page:** 1
**Author:** ![tareqmy](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/tareqmy/32/3946_2.png) [@tareqmy](https://community.asterisk.org/u/tareqmy)\ **Post date:** [February 4, 2018, 3:35pm UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/1 "2018-02-04T15:35:35Z")
Hi, I am running asterisk in docker exposing 5060 port and 10000-10010 for rtp. The docker is running in a host in local net 10.0.0.201 and the sip phones are also in 10.0.0.X. I have the transport like this — [transport-udp] type=transport protocol=udp bind=0.0.0.0 local\_net=172.22.0.2/16 external\_media\_address=10.0.0.201 external\_signaling\_address=10.0.0.201 I can register from the phone. But media fails to be established. What should i do? Thanks. ---
**Author:** ![tareqmy](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/tareqmy/32/3946_2.png) [@tareqmy](https://community.asterisk.org/u/tareqmy)\ **Post date:** [February 5, 2018, 6:36am UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/2 "2018-02-05T06:36:34Z")
I hope i have provided enough information for this. If any more information is needed please let me know. Please help. ---
**Author:** ![gjoseph](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/gjoseph/32/1690_2.png) [@gjoseph](https://community.asterisk.org/u/gjoseph)\ **Post date:** [February 5, 2018, 1:23pm UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/3 "2018-02-05T13:23:29Z")
Have you configured Asterisk and the client to restrict media to that port range? In Asterisk, you configure that in rtp.conf. Can you get a packet capture of the INVITE transaction? You can do that with wireshark or in the Asterisk CLI with “pjsip set logger on”. ---
**Author:** ![tareqmy](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/tareqmy/32/3946_2.png) [@tareqmy](https://community.asterisk.org/u/tareqmy)\ **Post date:** [February 13, 2018, 2:55pm UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/4 "2018-02-13T14:55:29Z")
Hi Joseph, First of all i apologise for the delayed response. Actually I thought nobody is replying after one day and didnt revisit the forum. Here is a more detailed conf and logs ---- docker-compose: > services: > asterisk15: > image: asterisk15:latest > container\_name: asterisk15 > hostname: asterisk15 > restart: unless-stopped > stdin\_open: true > ports: > - “5060:5060/udp” > - “10000-10010:10000-10010/udp” > networks: > astnet: > ipv4\_address: 192.168.10.4 NOTE: the docker network type is the default “bridge” In the rtp.conf > [general] > rtpstart=10000 > rtpend=10010 the transport in pjsip.conf > [transport-udp] > type=transport > protocol=udp > bind=0.0.0.0 > local\_net=192.168.10.0/8 > external\_media\_address=10.0.0.201 > external\_signaling\_address=10.0.0.201 The docker in running on host 10.0.0.201. The sip phone 1001 and 1002 are also running on 10.0.0.201 Here is the log after setting “pjsip set logger on” > asterisk15\*CLI\> pjsip set logger on > PJSIP Logging enabled > \<— Received SIP request (758 bytes) from UDP:192.168.0.1:53921 —\> > INVITE [sip:1002@10.0.0.201](mailto:sip:1002@10.0.0.201) SIP/2.0 > Via: SIP/2.0/UDP 10.0.0.201:60390;branch=z9hG4bK-524287-1—fea2d47645b08c14;rport > Max-Forwards: 70 > Contact: sip:1001@192.168.0.1:55566;rinstance=99b3721656576ad1 > To: sip:1002@10.0.0.201 > From: sip:1001@10.0.0.201;tag=023f0c04 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > CSeq: 1 INVITE > Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE > Content-Type: application/sdp > Supported: replaces > User-Agent: Bria 5 release 5.1.0 stamp 89373 > Content-Length: 206 > > v=0 > o=- 1518533195959320 1 IN IP4 10.0.0.201 > s=Bria 5 release 5.1.0 stamp 89373 > c=IN IP4 10.0.0.201 > t=0 0 > m=audio 62548 RTP/AVP 8 0 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > \<— Transmitting SIP response (519 bytes) to UDP:192.168.0.1:53921 —\> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 10.0.0.201:60390;rport=53921;received=192.168.0.1;branch=z9hG4bK-524287-1—fea2d47645b08c14 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > From: sip:1001@10.0.0.201;tag=023f0c04 > To: sip:1002@10.0.0.201;tag=z9hG4bK-524287-1—fea2d47645b08c14 > CSeq: 1 INVITE > WWW-Authenticate: Digest realm=“asterisk”,nonce=“1518533205/26e2dfc6d843fcf5727a89b28e975746”,opaque=“5465cbb73307e332”,algorithm=md5,qop=“auth” > Server: Asterisk PBX 15.2.0 > Content-Length: 0 > > \<— Received SIP request (337 bytes) from UDP:192.168.0.1:53921 —\> > ACK [sip:1002@10.0.0.201](mailto:sip:1002@10.0.0.201) SIP/2.0 > Via: SIP/2.0/UDP 10.0.0.201:60390;branch=z9hG4bK-524287-1—fea2d47645b08c14;rport > Max-Forwards: 70 > To: sip:1002@10.0.0.201;tag=z9hG4bK-524287-1—fea2d47645b08c14 > From: sip:1001@10.0.0.201;tag=023f0c04 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > CSeq: 1 ACK > Content-Length: 0 > > \<— Received SIP request (1039 bytes) from UDP:192.168.0.1:53921 —\> > INVITE [sip:1002@10.0.0.201](mailto:sip:1002@10.0.0.201) SIP/2.0 > Via: SIP/2.0/UDP 10.0.0.201:60390;branch=z9hG4bK-524287-1—8a39c1718a84a123;rport > Max-Forwards: 70 > Contact: sip:1001@192.168.0.1:55566;rinstance=99b3721656576ad1 > To: sip:1002@10.0.0.201 > From: sip:1001@10.0.0.201;tag=023f0c04 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > CSeq: 2 INVITE > Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE > Content-Type: application/sdp > Supported: replaces > User-Agent: Bria 5 release 5.1.0 stamp 89373 > Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“1518533205/26e2dfc6d843fcf5727a89b28e975746”,uri="sip:1002@10.0.0.201",response=“82cdca82a444dca5016c4b4a1798dca3”,cnonce=“336a448fd2e8ccf6b179f663f5fde342”,nc=00000001,qop=auth,algorithm=md5,opaque=“5465cbb73307e332” > Content-Length: 206 > > v=0 > o=- 1518533195959320 1 IN IP4 10.0.0.201 > s=Bria 5 release 5.1.0 stamp 89373 > c=IN IP4 10.0.0.201 > t=0 0 > m=audio 62548 RTP/AVP 8 0 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > == Setting global variable ‘SIPDOMAIN’ to ‘10.0.0.201’ > \<— Transmitting SIP response (326 bytes) to UDP:192.168.0.1:53921 —\> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.0.0.201:60390;rport=53921;received=192.168.0.1;branch=z9hG4bK-524287-1—8a39c1718a84a123 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > From: sip:1001@10.0.0.201;tag=023f0c04 > To: sip:1002@10.0.0.201 > CSeq: 2 INVITE > Server: Asterisk PBX 15.2.0 > Content-Length: 0 > > ``` > -- Executing [1002@local:1] NoOp("PJSIP/1001-00000000", "") in new stack > -- Executing [1002@local:2] Verbose("PJSIP/1001-00000000", "1, ""Jon Doe" <1001> dialed."") in new stack > > ``` > > ““Jon Doe” \<1001\> dialed.” > – Executing [1002@local:3] Set(“PJSIP/1001-00000000”, “CDR\_PROP(disable)=1”) in new stack > – Executing [1002@local:4] Goto(“PJSIP/1001-00000000”, “internal-main,1002,1”) in new stack > – Goto (internal-main,1002,1) > – Executing [1002@internal-main:1] Verbose(“PJSIP/1001-00000000”, “1, “User 1001 dialed 1002.””) in new stack > “User 1001 dialed 1002.” > – Executing [1002@internal-main:2] Set(“PJSIP/1001-00000000”, “DIALED\_EXTEN=1002”) in new stack > – Executing [1002@internal-main:3] GotoIf(“PJSIP/1001-00000000”, “0?dialed-BUSY,1:”) in new stack > – Executing [1002@internal-main:4] Dial(“PJSIP/1001-00000000”, “PJSIP/1002,30”) in new stack > – Called PJSIP/1002 > \<— Transmitting SIP request (949 bytes) to UDP:192.168.0.1:40589 —\> > INVITE sip:1002@192.168.0.1:40589;rinstance=a55a21f0a2d53ce2 SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.4:5060;rport;branch=z9hG4bKPj251d9796-937d-41ee-a521-7d89a4db74a3 > From: “Jon Doe” sip:1001@192.168.10.4;tag=50f8e875-39a0-40c9-8e2d-f5e662cf3e44 > To: sip:1002@192.168.0.1;rinstance=a55a21f0a2d53ce2 > Contact: sip:asterisk@192.168.10.4:5060 > Call-ID: 7dd3291f-e703-4ab0-a954-9e4d4a138e32 > CSeq: 6264 INVITE > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER > Supported: 100rel, timer, replaces, norefersub > Session-Expires: 1800 > Min-SE: 90 > Max-Forwards: 70 > User-Agent: Asterisk PBX 15.2.0 > Content-Type: application/sdp > Content-Length: 237 > > v=0 > o=- 1401251911 1401251911 IN IP4 192.168.10.4 > s=Asterisk > c=IN IP4 192.168.10.4 > t=0 0 > m=audio 10008 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv > > \<— Received SIP response (364 bytes) from UDP:192.168.0.1:40589 —\> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.10.4:5060;rport=5060;branch=z9hG4bKPj251d9796-937d-41ee-a521-7d89a4db74a3;received=10.0.0.201 > To: sip:1002@192.168.0.1;rinstance=a55a21f0a2d53ce2 > From: “Jon Doe” sip:1001@192.168.10.4;tag=50f8e875-39a0-40c9-8e2d-f5e662cf3e44 > Call-ID: 7dd3291f-e703-4ab0-a954-9e4d4a138e32 > CSeq: 6264 INVITE > Content-Length: 0 > > \<— Received SIP response (488 bytes) from UDP:192.168.0.1:40589 —\> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.10.4:5060;rport=5060;branch=z9hG4bKPj251d9796-937d-41ee-a521-7d89a4db74a3;received=10.0.0.201 > Contact: sip:1002@192.168.0.1:40589 > To: sip:1002@192.168.0.1;rinstance=a55a21f0a2d53ce2;tag=ae119224 > From: "Jon Doe"sip:1001@192.168.10.4;tag=50f8e875-39a0-40c9-8e2d-f5e662cf3e44 > Call-ID: 7dd3291f-e703-4ab0-a954-9e4d4a138e32 > CSeq: 6264 INVITE > User-Agent: Bria 3 release 3.5.5 stamp 71243 > Allow-Events: hold, talk > Content-Length: 0 > > ``` > -- PJSIP/1002-00000001 is ringing > > ``` > > \<— Transmitting SIP response (513 bytes) to UDP:192.168.0.1:53921 —\> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.0.0.201:60390;rport=53921;received=192.168.0.1;branch=z9hG4bK-524287-1—8a39c1718a84a123 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > From: sip:1001@10.0.0.201;tag=023f0c04 > To: sip:1002@10.0.0.201;tag=03e33564-5d15-4e9e-bbc8-b11df6b51c9f > CSeq: 2 INVITE > Server: Asterisk PBX 15.2.0 > Contact: sip:192.168.10.4:5060 > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER > Content-Length: 0 > > \<— Received SIP response (809 bytes) from UDP:192.168.0.1:40589 —\> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.10.4:5060;rport=5060;branch=z9hG4bKPj251d9796-937d-41ee-a521-7d89a4db74a3;received=10.0.0.201 > Contact: sip:1002@192.168.0.1:40589 > To: sip:1002@192.168.0.1;rinstance=a55a21f0a2d53ce2;tag=ae119224 > From: "Jon Doe"sip:1001@192.168.10.4;tag=50f8e875-39a0-40c9-8e2d-f5e662cf3e44 > Call-ID: 7dd3291f-e703-4ab0-a954-9e4d4a138e32 > CSeq: 6264 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces, eventlist > User-Agent: Bria 3 release 3.5.5 stamp 71243 > Content-Length: 204 > > v=0 > o=- 1518533199140469 3 IN IP4 10.0.0.201 > s=Bria 3 release 3.5.5 stamp 71243 > c=IN IP4 10.0.0.201 > t=0 0 > m=audio 52366 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > ``` > > 0x7fef4006f180 -- Strict RTP learning after remote address set to: 10.0.0.201:52366 > > ``` > > \<— Transmitting SIP request (421 bytes) to UDP:192.168.0.1:40589 —\> > ACK sip:1002@192.168.0.1:40589 SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.4:5060;rport;branch=z9hG4bKPje4c7318a-1f3b-43bf-a90c-a9782bbd03c5 > From: “Jon Doe” sip:1001@192.168.10.4;tag=50f8e875-39a0-40c9-8e2d-f5e662cf3e44 > To: sip:1002@192.168.0.1;rinstance=a55a21f0a2d53ce2;tag=ae119224 > Call-ID: 7dd3291f-e703-4ab0-a954-9e4d4a138e32 > CSeq: 6264 ACK > Max-Forwards: 70 > User-Agent: Asterisk PBX 15.2.0 > Content-Length: 0 > > ``` > -- PJSIP/1002-00000001 answered PJSIP/1001-00000000 > > 0x7fef40060a40 -- Strict RTP learning after remote address set to: 10.0.0.201:62548 > > ``` > > \<— Transmitting SIP response (817 bytes) to UDP:192.168.0.1:53921 —\> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.0.0.201:60390;rport=53921;received=192.168.0.1;branch=z9hG4bK-524287-1—8a39c1718a84a123 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > From: sip:1001@10.0.0.201;tag=023f0c04 > To: sip:1002@10.0.0.201;tag=03e33564-5d15-4e9e-bbc8-b11df6b51c9f > CSeq: 2 INVITE > Server: Asterisk PBX 15.2.0 > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER > Contact: sip:192.168.10.4:5060 > Supported: 100rel, timer, replaces, norefersub > Content-Type: application/sdp > Content-Length: 227 > > v=0 > o=- 263818264 3 IN IP4 192.168.10.4 > s=Asterisk > c=IN IP4 192.168.10.4 > t=0 0 > m=audio 10006 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv > > ``` > -- Channel PJSIP/1002-00000001 joined 'simple_bridge' basic-bridge <2f67b2aa-8d9f-4d19-859d-d5196246e5bd> > -- Channel PJSIP/1001-00000000 joined 'simple_bridge' basic-bridge <2f67b2aa-8d9f-4d19-859d-d5196246e5bd> > > Bridge 2f67b2aa-8d9f-4d19-859d-d5196246e5bd: switching from simple_bridge technology to native_rtp > > Locally RTP bridged 'PJSIP/1001-00000000' and 'PJSIP/1002-00000001' in stack > > ``` > > \<— Transmitting SIP response (817 bytes) to UDP:192.168.0.1:53921 —\> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.0.0.201:60390;rport=53921;received=192.168.0.1;branch=z9hG4bK-524287-1—8a39c1718a84a123 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > From: sip:1001@10.0.0.201;tag=023f0c04 > To: sip:1002@10.0.0.201;tag=03e33564-5d15-4e9e-bbc8-b11df6b51c9f > CSeq: 2 INVITE > Server: Asterisk PBX 15.2.0 > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER > Contact: sip:192.168.10.4:5060 > Supported: 100rel, timer, replaces, norefersub > Content-Type: application/sdp > Content-Length: 227 > > v=0 > o=- 263818264 3 IN IP4 192.168.10.4 > s=Asterisk > c=IN IP4 192.168.10.4 > t=0 0 > m=audio 10006 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv > > \<— Transmitting SIP response (817 bytes) to UDP:192.168.0.1:53921 —\> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.0.0.201:60390;rport=53921;received=192.168.0.1;branch=z9hG4bK-524287-1—8a39c1718a84a123 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > From: sip:1001@10.0.0.201;tag=023f0c04 > To: sip:1002@10.0.0.201;tag=03e33564-5d15-4e9e-bbc8-b11df6b51c9f > CSeq: 2 INVITE > Server: Asterisk PBX 15.2.0 > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER > Contact: sip:192.168.10.4:5060 > Supported: 100rel, timer, replaces, norefersub > Content-Type: application/sdp > Content-Length: 227 > > v=0 > o=- 263818264 3 IN IP4 192.168.10.4 > s=Asterisk > c=IN IP4 192.168.10.4 > t=0 0 > m=audio 10006 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv NOTE: When there is no rtp detected by asterisk after some time it sends BYE but does not get any response – > \<— Transmitting SIP request (421 bytes) to UDP:192.168.0.1:55566 —\> > BYE sip:1001@192.168.0.1:55566;rinstance=99b3721656576ad1 SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.4:5060;rport;branch=z9hG4bKPjfd47ce99-150e-490b-8a97-082d99ff7c1e > From: sip:1002@10.0.0.201;tag=03e33564-5d15-4e9e-bbc8-b11df6b51c9f > To: sip:1001@10.0.0.201;tag=023f0c04 > Call-ID: 89373MGMxOGYxNDYyMTA5Mjg3MTdlMWNkZWVmZGY2NDEyM2Q > CSeq: 26006 BYE > Max-Forwards: 70 > User-Agent: Asterisk PBX 15.2.0 > Content-Length: 0 If any more details is needed please let me know. Thanks in advance. ---
**Author:** ![gjoseph](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/gjoseph/32/1690_2.png) [@gjoseph](https://community.asterisk.org/u/gjoseph)\ **Post date:** [February 13, 2018, 3:31pm UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/5 "2018-02-13T15:31:44Z")
I’m a little confused by your network setup. You say that the docker host is 10.0.0.201 and the docker network is “bridged” but you’ve given your container a 192.168.10.4 address so there’s no natting. The transport has local\_net defined as 192.168.10.0/8 which will include all 192.x.x.x addresses. Requests from the Bria phone for extension 1001 are coming from 192.168.0.1 but reporting 10.0.0.201 in the SDP. Requests to extension 1002 are also going to 192.168.0.1 So, why the 192 network at all? I’d pick one network and get it working, then if you need more than one, we can go from there. ---
**Author:** ![tareqmy](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/tareqmy/32/3946_2.png) [@tareqmy](https://community.asterisk.org/u/tareqmy)\ **Post date:** [February 13, 2018, 3:40pm UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/6 "2018-02-13T15:40:32Z")
ok my apologies for the network masking. it should be 192.168.10.0/24. the docker is run on a machine which is the host “10.0.0.201”. But when docker is deployed, it has its own private network which is 192.168.10.0/24. If i run the docker in network:host mode it works fine without the extra nat config in transport section. But i wanted to avoid running docker in host mode. And after i have corrected it, now it is working. Thanks Joseph. ---
**Author:** ![fukemy](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/fukemy/32/7658_2.png) [@fukemy](https://community.asterisk.org/u/fukemy)\ **Post date:** [April 11, 2024, 7:14am UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/7 "2024-04-11T07:14:18Z")
> [@tareqmy](#): > > But i wanted to avoid running docker in host mode. Hi, can you provide me the way u solved this? THanks so much, i am getting same issue ---
**Author:** ![Vladislav](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/vladislav/32/7553_2.png) [@Vladislav](https://community.asterisk.org/u/Vladislav)\ **Post date:** [May 22, 2024, 9:28am UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/8 "2024-05-22T09:28:50Z")
Hi Could you please provide any clarification on what exactly is helping you. Because I faced the same problem and all the discussions above did not solve my problem. ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [May 31, 2024, 12:05pm UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/9 "2024-05-31T12:05:58Z")
5 posts were merged into an existing topic: [Unable to establishing a call](https://community.asterisk.org/t/unable-to-establishing-a-call/102560/7) ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [May 31, 2024, 12:06pm UTC](https://community.asterisk.org/t/asterisk-within-docker/73412/10 "2024-05-31T12:06:12Z")