The DSM 6 release notes say this is no longer supported by the open source team, so Synology is not supporting it in DSM 6. It will not run after the upgrade. I had to uninstall this package in order to run the DSM 6 upgrade. It wouldn’t shut down and the upgrade wouldn’t proceed with it running.
I wonder if anyone has any insight working with Asterisk on a Synology nas. I'm trying to set up inbound and outbound calling but am having problems. My main problem is getting inbound calls to route to an extension. Local switching between extensions is working fine (physical IP handsets), and my trunks are showing as registered with Sipgate, but when an external call hits the Asterisk box it doesn't go anywhere regardless of what inbound routes I try - have tried using a 'catch all' route and also a specific route with my number in the DID, but the calls fail to connect to an extension and I can't see any error messages anywhere.
Asterisk/1.8.13.1 Asterisk GUI-version : 2.1.0-rc1 Sip dialog below, can anyone see any obvious problems? (numbers have been altered slightly to protect the innocent;) <--- SIP read from UDP:217.10.79.23:5060 ---> INVITE sip:[email protected]:5060 SIP/2.0 Record-Route: <sip:217.10.79.23:5060;lr;ftag=as0785e8b8> Record-Route: <sip:172.20.40.3;lr=on> Record-Route: <sip:217.10.79.23:5060;lr;ftag=as0785e8b8> Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bK46f5.6b541091.0 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK46f5.6b541091.0 Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.222;branch=z9hG4bK15e50793 Via: SIP/2.0/UDP 217.116.117.10:5060;received=217.116.117.10;branch=z9hG4bK15e50793;rport=5060 Max-Forwards: 67 From: '07957123456' <sip:[email protected]>;tag=as0785e8b8 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 467 v=0 o=root 759523207 759523207 IN IP4 217.116.117.10 s=sipgate VoIP GW c=IN IP4 217.10.77.244 t=0 0 m=audio 58172 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=direction:active a=nortpproxy:yes <-------------> --- (18 headers 21 lines) --- Sending to 217.10.79.23:5060 (NAT) Using INVITE request as basis request - [email protected] Found peer '4153553' for '07957123456' from 217.10.79.23:5060 <--- Reliably Transmitting (NAT) to 217.10.79.23:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bK46f5.6b541091.0;received=217.10.79.23;rport=5060 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK46f5.6b541091.0 Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.222;branch=z9hG4bK15e50793 Via: SIP/2.0/UDP 217.116.117.10:5060;received=217.116.117.10;branch=z9hG4bK15e50793;rport=5060 From: '07957123456' <sip:[email protected]>;tag=as0785e8b8 To: <sip:[email protected]>;tag=as392a975a Call-ID: [email protected] CSeq: 102 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm='asterisk', nonce='7bdf4623' Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) <--- SIP read from UDP:217.10.79.23:5060 ---> ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bK46f5.6b541091.0 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK46f5.6b541091.0 From: '07957123456' <sip:[email protected]>;tag=as0785e8b8 Call-ID: [email protected] To: <sip:[email protected]>;tag=as392a975a CSeq: 102 ACK Max-Forwards: 69 Content-Length: 0 X-hint: rr-enforcedHello there fellow Asterisk users and developers. From start I have to mention I am new to Asterisk, but I am trying to learn as much as possible. What do I have I need to deploy in our office 11 VoIP phones by Cisco - SPA502G. We have a NAS by Synology - DS1511+ that has Asterisk installed as a package. We have a SIP account from our provider. Asterisk version is 1.8.13.1, Asterisk GUI-version : 2.1.0-rc1 Cisco VoIP phones updated to the latest version available. What do I need Setup the SIP account on Asterisk server Setup the user extension ![]() What I did I have installed Asterisk package and I am able to access it The server on which Asterisk resides has two network cards. One NIC is assigned a public IP and is connected directly to the internet, without firewall. One NIC is assigned a private IP and is connected to the local network, behind NAT and firewall. The Cisco VoIP phones are connected to the local network and I can access their web server What is not working While setting up the SIP account from our provider, I get the Status 'Requesting' written in red and It does not register. Our provider stated that their platform has problems with Asterisk 1.8 and we should setup the SIP account without user name and password; only host ![]() fullname=TestUser1 registersip=no host=dynamic callgroup=1 mailbox=203 call-limit=100 type=peer username=203 transfer=yes callcounter=yes context=DLPN_DialPlan1 cid_number=203 hasvoicemail=yes vmsecret=test email= threewaycalling=no hasdirectory=no callwaiting=no hasmanager=no hasagent=no hassip=yes hasiax=no secret=test nat=yes canreinvite=no dtmfmode=rfc2833 insecure=no pickupgroup=1 disallow=all allow=ulaw,gsm macaddress=203 autoprov=yes label=203 linenumber=1 LINEKEYS=1 I have setup the account on the Cisco SPA502G phone, but I constantly get error 404 (on the phone) I tried various settings and options, but all lead to the dreaded 404 error. The questions 1. What might be causing this 404 error and how can I fix it? 2. If the SIP account is setup on the Asterisk box but no phones are setup yet, when I call the SIP number, should I hear the line ringing on the calling side? At the moment I do not hear any rings and the call just drops, without any notice (busy, etc) sip.conf context=default allowoverlap=yes udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 transport=udp srvlookup=yes subscribecontext=default allowexternaldomains=yes allowguest=yes allowsubscribe=yes allowtransfer=yes alwaysauthreject=no autodomain=yes bindaddr=0.0.0.0 bindport=5060 callevents=no checkmwi=10 compactheaders=no defaultexpiry=120 dumphistory=no externrefresh=10 fromdomain=domain.com g726nonstandard=no jbenable=no jbforce=no jblog=no maxcallbitrate=384 maxexpiry=3600 minexpiry=60 mohinterpret=default notifyringing=yes pedantic=no progressinband=never promiscredir=no realm=domain.com recordhistory=yes registerattempts=0 registertimeout=20 relaxdtmf=no sendrpid=no sipdebug=yes t1min=100 t38pt_udptl=no tos_audio=none tos_sip=none tos_video=none trustrpid=no useragent=Voice Server usereqphone=no videosupport=no dtmfmode=rfc2833 nat=no externip=XX.XX.XX.226 localnet=10.70.1.0/255.255.0.0 allow=ulaw,alaw,gsm,ilbc,speex,g726,adpcm,lpc10,g729,g723,h263,h263p,h264 Thank you very much Comments are closed.
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