Lonnie, Below is my Asterisk CLI output with “sip set debug on”; it is a call from my mobile to 1 of the 2 DIDs I have registered with my ITSP. In my reading of the output, it appears that even though I’ve called 705-797-4986, it is routed to me via 705-797-4933.
Really scratching my head on this one… would really appreciate any pointers you can give! My SIP.conf includes: [general] context=default register => 7057974933:[email protected]/7057974933 register => 7057974986:[email protected]/7057974986 [testtemplate](!) type=peer call-limit=5 disallow=all allow=ulaw qualify=yes [7057974933](testtemplate) ; incoming calls host=sip05.unlimitel.ca fromuser=7057974933 remotesecret=PASSWORD1 canreinvite=no [7057974986](testtemplate) ; incoming calls host=sip05.unlimitel.ca fromuser=7057974986 remotesecret=PASSWORD2 canreinvite=no <--- SIP read from UDP:209.217.98.154:5060 ---> INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK57d6f0e1;rport Max-Forwards: 70 From: "6139929344" <sip:[email protected]>;tag=as65cd906e To: <sip:[email protected]:5060> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Primus-Unlimitel Date: Tue, 27 May 2014 00:37:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 259 v=0 o=root 1448758994 1448758994 IN IP4 209.217.98.154 s=Primus-Unlimitel c=IN IP4 209.217.98.154 t=0 0 m=audio 17652 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to 209.217.98.154:5060 (no NAT) Sending to 209.217.98.154:5060 (no NAT) Using INVITE request as basis request - [email protected] Found peer '7057974933' for '6139929344' from 209.217.98.154:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw), peer - audio=(gsm|ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 209.217.98.154:17652 Looking for 7057974986 in default (domain 198.72.123.133) list_route: hop: <sip:[email protected]> <--- Transmitting (no NAT) to 209.217.98.154:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK57d6f0e1;received=209.217.98.154;rport=5060 From: "6139929344" <sip:[email protected]>;tag=as65cd906e To: <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 102 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:[email protected]:5060> Content-Length: 0 <------------> -- Executing [7057974986@default:1] Answer("SIP/7057974933-0000000b", "") in new stack Audio is at 25012 Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 209.217.98.154:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK57d6f0e1;received=209.217.98.154;rport=5060 From: "6139929344" <sip:[email protected]>;tag=as65cd906e To: <sip:[email protected]:5060>;tag=as4a7817dc Call-ID: [email protected] CSeq: 102 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:[email protected]:5060> Content-Type: application/sdp Require: timer Content-Length: 253 v=0 o=root 1177512703 1177512703 IN IP4 198.72.123.133 s=Asterisk PBX 11.7.0~dfsg-1ubuntu1 c=IN IP4 198.72.123.133 t=0 0 m=audio 25012 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:209.217.98.154:5060 ---> ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK4b7802d3;rport Max-Forwards: 70 From: "6139929344" <sip:[email protected]>;tag=as65cd906e To: <sip:[email protected]:5060>;tag=as4a7817dc Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Primus-Unlimitel Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Executing [7057974986@default:2] Echo("SIP/7057974933-0000000b", "") in new stack <--- SIP read from UDP:209.217.98.154:5060 ---> BYE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK5d991769;rport Max-Forwards: 70 From: "6139929344" <sip:[email protected]>;tag=as65cd906e To: <sip:[email protected]:5060>;tag=as4a7817dc Call-ID: [email protected] CSeq: 103 BYE User-Agent: Primus-Unlimitel X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 209.217.98.154:5060 (no NAT) Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 209.217.98.154:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.217.98.154:5060;branch=z9hG4bK5d991769;received=209.217.98.154;rport=5060 From: "6139929344" <sip:[email protected]>;tag=as65cd906e To: <sip:[email protected]:5060>;tag=as4a7817dc Call-ID: [email protected] CSeq: 103 BYE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (default, 7057974986, 2) exited non-zero on 'SIP/7057974933-0000000b' > Shamus, > > 'type=peer' is what you want for each DID's sip context. You probably need > to register for each DID as well for your situation. > > Lonnie > >> On 2014 May 26, at 10:50, Shamus Rask <[email protected]> wrote: >> >>> Lonnie, >>> >>> Yes, I should have specified that each DID has it’s own user/password (user >>> is just the DID number). Should I be looking at a type= other than peer? Am >>> I looking down the right track? >>> >>> >>> many thanks, >>> Shamus >>> >>> >>>> Message: 5 >>>> Date: Mon, 26 May 2014 09:27:57 -0500 >>>> From: Lonnie Abelbeck <[email protected]> >>>> Subject: Re: [Astlinux-users] help with multiple DID/SIP trunks >>>> To: AstLinux Users Mailing List <[email protected]> >>>> Message-ID: <[email protected]> >>>> Content-Type: text/plain; charset=windows-1252 >>>> >>>> Hi Shamus, >>>> >>>> Does the unlimitel.ca service allow sub-accounts for DID's, such that >>>> unique user/pass credentials could be specified for each DID. This >>>> requires additional configuration in sip.conf but should get you around >>>> the aggregated 5 channel limit. >>>> >>>> You might also inquire why the 5 channel limit exists, sometimes that is a >>>> default sanity setting by the SIP provider and can be raised at no >>>> additional cost. >>>> >>>> Lonnie >>>> >>>> >>>> On May 26, 2014, at 8:30 AM, Shamus Rask wrote: >>>> >>>>> I?m posting here as I find the AstLinux community to be the most friendly >>>>> and knowledgeable about all things Asterisk! >>>>> >>>>> My ITSP offers DID/SIP trunks at a very competitive rate?each DID >>>>> includes 5 channels. I?m running Asterisk 11. >>>>> >>>>> The ITSP only offers a single server for both incoming and outgoing >>>>> calls: sip05.unlimitel.ca. I recently discovered that with the >>>>> configuration I had, all of my calls, no matter how many DIDs I have, >>>>> were being sent over a single trunk (the first to register from >>>>> Asterisk). I believe this is due to my selecting type=peer in my >>>>> SIP.conf; it appears to match based on IP & port so all of the DID/trunks >>>>> appear as a single one. This means that I?m limited to 5 channels, >>>>> despite having 4xDIDs which should give me 20 channels (5 per DID). >>>>> >>>>> I?ve tried changing this to type=user, but although I see registration >>>>> (sip show registry) and users (sip show users) I cannot see any incoming >>>>> calls. >>>>> >>>>> Can anyone offer any help/suggestions? Pulling my hair out! >>>>> Shamus >>>> >>>> >>>> >>>> >>>> ------------------------------ >>>> >>>> ------------------------------------------------------------------------------ >>>> The best possible search technologies are now affordable for all companies. >>>> Download your FREE open source Enterprise Search Engine today! >>>> Our experts will assist you in its installation for $59/mo, no commitment. >>>> Test it for FREE on our Cloud platform anytime! >>>> http://pubads.g.doubleclick.net/gampad/clk?id=145328191&iu=/4140/ostg.clktrk >>>> >>>> ------------------------------ >>>> >>>> _______________________________________________ >>>> Astlinux-users mailing list >>>> [email protected] >>>> https://lists.sourceforge.net/lists/listinfo/astlinux-users >>>> >>>> Donations to support AstLinux are graciously accepted via PayPal to >>>> [email protected]. >>>> >>>> End of Astlinux-users Digest, Vol 94, Issue 10 >>>> ********************************************** >>> >> ------------------------------------------------------------------------------ The best possible search technologies are now affordable for all companies. 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