Ioan, Thanks for the info… I’ve tried this configuration and I still reach a hard limit of 5 simultaneous calls. It appears my ITSP has put some sort of call-limit=5, and since Asterisk is not routing the calls properly, I reach it despite having more than one DID.
During my testing, I was using Asterisk 1.8 on AstLinux. At this point, I was actually able to uniquely route the calls with type=user in SIP.conf. I then had to have a different Dial() statement in extensions.conf as type=user only accepts incoming calls, but I did get it all working. Sadly, this stopped working when I moved to Asterisk 11… Here is what I had working on Asterisk 1.8… would be interested in anyone else’s feedback or suggestions as to why this no longer works on Asterisk 11: extensions.conf ——— Dial(SIP/NUMBERTOBECALLED:PASSWORD1::[email protected]) sip.conf ——— register => 6139929999:[email protected]/6139929999 register => 6139929998:[email protected]/6139929998 [6139929999] ; incoming calls type=user host=sip06.unlimitel.ca call-limit=5 canreinvite=no qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 [6139929998] ; incoming calls type=user host=sip06.unlimitel.ca call-limit=5 canreinvite=no qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 > ------------------------------ > > Message: 4 > Date: Tue, 27 May 2014 08:19:34 +0300 > From: Ioan Indreias <[email protected]> > Subject: Re: [Astlinux-users] help with multiple DID/SIP trunks > To: [email protected] > Message-ID: <[email protected]> > Content-Type: text/plain; charset="windows-1252" > > Hello, > > From my experience you could not make any improvements as this is an > Asterisk well known bug: in case you have multiple SIP accounts defined > to the same IP/hostname Asterisk will match the call to the first one it > will find in the configuration file (I think you will find more > information on this subject if you'll search the archive of the main > Asterisk list). > > Nevertheless only the channel name is affected by this issue - the > routing it is OK. > > As you could see: > > Found peer '*7057974933*' for '6139929344' from *209.217.98.154:5060* > Executing [*7057974986*@default:2] Echo("*SIP/7057974933-0000000b*", > "") in new stack > > Green = DID = OK > Red = Channel Name = Wrong info > > In conclusion you should adapt your dialplan to overwrite some CDR > information (like accountcode) in order to be able to trace/bill > correctly the call to the right DID. > > HTH, > Ioan > www.modulo.ro > > On 27-May-14 4:23 AM, Shamus Rask wrote: >> 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. >> 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]. > > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > ------------------------------------------------------------------------------ > 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 11 > ********************************************** ------------------------------------------------------------------------------ Time is money. 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