[OpenSIPS-Users] opensips+dispatcher+asterisk problem

Tue Jun 30 13:50:18 CEST 2009

Hi Ram,

I found your email on the Asterisk mailing list also ;)

So, to answer here also: do you get any reply back from Asterisk ?

Regards,
Bogdan

ram wrote:
> Hi all
>
> After a long iam back to forum
>  
> back to my own topic and several readings done on this forum
> how people doing same kind of setup what iam trying to achive
>
> so here i have done some good developements
>
> for testing iam doing all in one Server
>
> Step1 :
>
> Installed in Fresh BOX with Debian
>
> Asterisk and A2B working Fine
>
>
> Step2 : registered with SIP account iam able to make calls successfully
>
> Step3 :
>
> installed Opensips
>
> Made Subscribers to view from A2b Database
>
> Step4 : changed Asterisk port from 5060 to 5062
>
> Step5 : Opensip config made changes to register users with Opensips
> and when they dial 001X call send to Asterisk box
>
>
> route[3]{
>
> if (uri =~ "sip:001[0-9]@*"){
> log(1, "Forwarding to Asterisk \n");
> rewritehostport("A2b-asterisk-IP:5062");
> route(1);
> exit;
> }
>
> Works Fine, No problems as of now
>
> But to go in advance, i want to use Number of * boxes to achive more Load
>
> Step5 : added Dispatcher Module in the Opensips
>
> loadmodule "dispatcher.so"
> .
> .
> .
> modparam("dispatcher","list_file","/usr/local/etc/opensips/dispatcher.cfg")
> .
> .
> .
> .
> changed route to use dispatcher
>
> route[3]{
>
> if (uri =~ "sip:001[0-9]@*"){
> log(1, "Forwarding to Asterisk \n");
> ds_select_dst("2","4");
> forward();
> route(1);
> exit;
> }
>
>
> My dispatcher Config Looks like below
>
> dispatcher.cfg
> 2 sip:a2b-asterisk-ip:5062
> 2 sip:a2b-asterisk-ip2:5062
>
> I have restarted Opensips
>
> when i dial 0017XXXXXX number the call send Opensips to Asterisk
>
>
>
> Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk
> Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2]
> Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg 
> hash [1]
> Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: 
> selected [4-2/1] <sip:a2b-asterisk-ip:5062>
> Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing 
> DNS lookup...
> Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request: 
> sending:#012INVITE sip:0017XXXXXXXX at opensips-ip:5060 
> SIP/2.0#015#012Record-Route: <sip:opensips-ip;lr=on>#015#012Via: 
> SIP/2.0/UDP opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: 
> SIP/2.0/UDP 
> ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From: 
> 4720779942 <sip:4720779942 at opensips-ip:5060>;tag=1966722825#015#012To: 
> 0017325824631 <sip:0017XXXXXXX at opensips-ip:5060>#015#012Call-ID: 
> 32167199575863-11502744529360 at ip-phoneip#015#012CSeq: 2 
> INVITE#015#012Contact: 
> <sip:4720779942 at ipphone-ip:5060>#015#012Proxy-Authorization: Digest 
> username="4720779942", realm="asterisk", nonce="79ee65ba", 
> uri="sip:0017XXXXXX at opensips-ip:5060", 
> response="3e182f165a5663d0b145d6b55d34e94b", 
> algorithm=MD5#015#012Max-Forwards: 69#015#012Supported: 
> replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK, 
> OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, 
> UPDATE#015#012Content-Type: application/sdp#015#012Content-Length: 
> 319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4 
> 202.63.111.2#015#012s=A conversation#015#012c=IN IP4 
> ip-phone-ip#015#012t=0 0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 
> 101#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:4 
> G723/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 
> PCMU/8000#015#012a=rtpmap:9 G722/16000#015#012a=rtpmap:101 
> telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012.
> opensips[25868]: DBG:core:forward_request: orig. len=1087, 
> new_len=1220, proto=1
>
>
>
> when i ngrep
> ------------
>
>
> U 2009/06/30 01:59:20.770599 ipphone:5060 -> asterisk-a2b-ip:5060
> INVITE sip:0017XXXXXXXX at asterisk-a2b-ip:5060 SIP/2.0.
> Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport.
> From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
> To: 0017XXXXXXXX <sip:0017XXXXXXXX at asterisk-a2b-ip:5060>.
> Call-ID: 14399316162240-7371067914582 at ipphone.
> CSeq: 2 INVITE.
> Contact: <sip:4720779942 at ipphone:5060>.
> Proxy-Authorization: Digest username="4720779942", realm="asterisk", 
> nonce="07ba8624", uri="sip:0017XXXXXXXX at asterisk-a2b-ip:5060", 
> response="5dbe9b2937d0bc3f6e8d25052fff0b6a", algorithm=MD5.
> Max-Forwards: 70.
> Supported: replaces.
> User-Agent: Voip Phone 1.0.
> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, 
> SUBSCRIBE, PRACK, UPDATE.
> Content-Type: application/sdp.
> Content-Length: 319.
> .
> v=0.
> o=4720779942 69102627 18481147 IN IP4 ipphone.
> s=A conversation.
> c=IN IP4 ipphone.
> t=0 0.
> m=audio 10034 RTP/AVP 18 4 8 0 9 101.
> a=rtpmap:18 G729/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:9 G722/16000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
>
>
> U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 -> ipphone:5060
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP 
> ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060.
> From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
> To: 0017XXXXXXXX <sip:0017XXXXXXXX at asterisk-a2b-ip:5060>.
> Call-ID: 14399316162240-7371067914582 at ipphone.
> CSeq: 2 INVITE.
> Server: OpenSIPS (1.5.1-notls (i386/linux)).
> Content-Length: 0.
> .
>
>
> U 2009/06/30 01:59:21.650498 asterisk-a2b-ip:5060 -> ipphone:5060
> SIP/2.0 407 Proxy Authentication Required.
> Via: SIP/2.0/UDP 
> ipphone:5060;received=ipphone;branch=z9hG4bK1984515716453028636;rport=5060.
> From: 4720779942 <sip:4720779942 at asterisk-a2b-ip:5060>;tag=3037030266.
> To: 0017XXXXXXXX <sip:0017XXXXXXXX at asterisk-a2b-ip:5060>;tag=as0cb075c5.
> Call-ID: 14399316162240-7371067914582 at ipphone.
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", 
> nonce="07ba8624".
> Content-Length: 0.
>
> ------
>
> when i enable debug at Asterisk and Look at i see the below error
> ---------------------------------------------------------------
>
> <--- SIP read from a2b-asterisk-ip:5060 --->
> INVITE sip:0017XXXXXXXXX at a2b-asterisk-ip:5060 SIP/2.0
> Record-Route: <sip:a2b-asterisk-ip;lr=on>
> Via: SIP/2.0/UDP a2b-asterisk-ip;branch=z9hG4bK166.1b7e2827.0
> Via: SIP/2.0/UDP 
> Ip-phone:5060;received=Ip-phone;branch=z9hG4bK295731884823024293;rport=5060
> From: 4720779942 <sip:4720779942 at a2b-asterisk-ip:5060>;tag=12544334
> To: 0017XXXXXXXXX <sip:0017XXXXXXXXX at a2b-asterisk-ip:5060>
> Call-ID: 16946271051109-143302828620026 at Ip-phone
> CSeq: 1 INVITE
> Contact: <sip:4720779942 at Ip-phone:5060>
> Max-Forwards: 69
> Supported: replaces
> User-Agent: Voip Phone 1.0
> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, 
> SUBSCRIBE, PRACK, UPDATE
> Content-Type: application/sdp
> Content-Length: 319
>
> v=0
> o=4720779942 31008195 22123120 IN IP4 Ip-phone
> s=A conversation
> c=IN IP4 Ip-phone
> t=0 0
> m=audio 10030 RTP/AVP 18 4 8 0 9 101
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
>
> <------------->
> [Jun 30 01:15:29] VERBOSE[24612] logger.c: --- (15 headers 14 lines) ---
> [Jun 30 01:15:29] VERBOSE[24612] logger.c: Ignoring this INVITE request
> [Jun 30 01:15:31] VERBOSE[24612] logger.c: Reliably Transmitting (no 
> NAT) to termination-provider-ip:5062:
> OPTIONS sip:termination-provider-ip:5062 SIP/2.0
> Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport
> From: "asterisk" <sip:asterisk at a2b-asterisk-ip:5062>;tag=as4cf91fd8
> To: <sip:termination-provider-ip:5062>
> Contact: <sip:asterisk at a2b-asterisk-ip:5062>
> Call-ID: 65a49c0977c6de0a1d2dbbfe757724bd at a2b-asterisk-ip
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Tue, 30 Jun 2009 08:15:31 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> [Jun 30 01:15:32] VERBOSE[24612] logger.c:
> <--- SIP read from termination-provider-ip:5062 --->
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport=5062
> From: "asterisk" <sip:asterisk at a2b-asterisk-ip:5062>;tag=as4cf91fd8
> To: 
> <sip:termination-provider-ip:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.ee5a
> Call-ID: 65a49c0977c6de0a1d2dbbfe757724bd at a2b-asterisk-ip
> CSeq: 102 OPTIONS
> Content-Length: 0
>
> ---------
>
>
> why does Asterisk sending with out any values
>
> ---
>
> From: "asterisk" <sip:asterisk at a2b-asterisk-ip:5062>;tag=as4cf91fd8
> To: <sip:termination-provider-ip:5062>
>
> ---
>
> Any suggestions
>
> Ram
> ------------------------------------------------------------------------
>
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