sr-users October 2011

sr-users@lists.kamailio.org

69 participants
117 discussions

How to limit maximum number of calls
by Austin Einter 28 Oct '11

28 Oct '11

Hi I am running kamailio proxy (1.5) as an intermediate proxy, where all my SIP signalling packets are passing through. I want to limit maximum number of calls kamailio proxy server can handle at a time. How can I do this, please give me some pointer in this regard. Regards Austin

4 7

new email contact
by Timo Reimann 28 Oct '11

28 Oct '11

Hey all, this is just to let everybody know that my email contact has changed to sr(a)foo-lounge.de . Of course, I'm still on the lists but if anyone aware of my former email address wishes to contact me directly, please refer to the new address in the future. Thanks a lot and cheers, --Timo

1 0

Asterisk 1.8 and Kamailio 1.5 issue
by Rowell Rufino 27 Oct '11

27 Oct '11

Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do. Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ------------------------------ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ------------------------------ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ------------------------------ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 *X-Asterisk-HangupCause: Protocol error, unspecified *X-Asterisk-HangupCauseCode: 111 Content-Length: 0 Regards, Rowell

2 1

Asterisk 1.8 and Kamailio 1.5 issue
by Rowell Rufino 26 Oct '11

26 Oct '11

Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do. Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ________________________________ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ________________________________ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ________________________________ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ________________________________ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 Regards, Rowell

1 0

Who is at Astricon?
by JR Richardson 26 Oct '11

26 Oct '11

I am. On 10/25/11, sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Dispatcher Confusion (v3.2.0) (Daniel-Constantin Mierla) > 2. Re: Dispatcher Confusion (v3.2.0) (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2011 05:44:06 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Dispatcher Confusion (v3.2.0) > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <4EA78206.4060203(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > On 10/25/11 5:52 PM, Asgaroth wrote: >> Hi Daniel, >> >> Thanks for the clarrification, >> >> On 25/10/2011 16:30, Daniel-Constantin Mierla wrote: >>> >>> >>> 4.6. |ds_mark_dst("s")| >>> >>> Mark the last used address from destination set as inactive >>> ("i"/"I"/"0"), active ("a"/"A"/"1") or probing ("p"/"P"/"2"). With >>> this function, an automatic detection of failed gateways can be >>> implemented. When an address is marked as inactive or probing, it >>> will be ignored by 'ds_select_dst' and 'ds_select_domain'. >>> >> >> Above is the part that is a little misleading, it says that >> >> "When an address is marked as inactive or probing, it will be ignored >> by 'ds_select_dst' and 'ds_select_domain'." >> >> This, to me, means that if a gateway is in probing mode >> (Active-Probing) then it wont be selected in the destination set >> because it is in probing mode, if this is not the case then maybe the >> above line should read: >> >> "When an address is marked as inactive or inactive-probing, it will be >> ignored by 'ds_select_dst' and 'ds_select_domain'." >> >> This brings me to my next question then, how would I set the state of >> a destination to Inactive-Probing (state: IP) from the routing script. >> The ds_mark_dst() function only allows one of "a", "i", "p". >> >> If I do a ds_mark_dst("i"), then the state changes to "IX", Inactive >> with no probing to tell when gateway is back up. >> >> If I do a ds_mark_dst("i") and then right after ds_mark_dst("p"), a >> log is printed saying that you cannot put a destination into probing >> state when it is marked as inactive. > are you sure you run the devel version? There is no such case in the > sources -- send me exactly the log message you get. Only when the > destination is disabled the probing cannot be set, but not the same case > of inactive. > >> >> Is it possible to set the state of a gateway to inactive-probing from >> the routing script? > > Yes, it should be, no constraint in master branch source code. > > Daniel > >> >> Thanks >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > >

2 1

sip-router listen address
by caio 25 Oct '11

25 Oct '11

Hello, Is there a problem or anything to consider when sharing the same ip/interface between sip proxy and asterisk? I have asterisk on port 5080 and SR at 5060 in the same interface with the same IP address and have a missing BYE when call is over. However, moving asterisk service to different interface and ip in the same box, the issue is solved. I appreciate your comments. Regards, Claudio

3 4

Kamailio.cfg file woes
by Peter Schrock 24 Oct '11

24 Oct '11

I am wondering two things. 1. is there a place on the web that I may find a basic cfg file to at least get Kamailio running. 2. is there a resource that tells you what to include for what you want to use the system for. At this point, it seems like I should just include everything because I don't know what I might not need. I looked at the references on the website and found the information a little overwhelming with all the options that are available. I think smaller bit sizes would be helpful at this time for me. Peter

3 3

onsend_route and replies
by Asgaroth 24 Oct '11

24 Oct '11

Hi All, Does the onsend route get executed for all messages? I'm just performing some xlog's in the onsend_route, but I'm not seeing it being called for replies. Currently my onsend_route looks like: onsend_route { xlog("onsend_route : $rm : Recieve Details $si:$sp -> $Ri:$Rp"); xlog("onsend_route : $rm : Destination Details $snd(af) $snd(proto):$snd(ip):$snd(port)"); } I can see it being executed for initial messages coming in but I dont see it executed for any replies. Thanks

1 1

kamctl dialplan addrule
by Arthur Alkhasyan 24 Oct '11

24 Oct '11

Hi all, I'm new to the list and new to the Kamailio. We are using Kamailio with Mysql to route calls between clients asterisks and SIP PSTN gateways. We have two inbound gateways and one outbound. I need to add outbound gateway and a few new features: 1. Call routing between carriers based at prefixes added by asterisks. I would like to route all calls with prefix 99101 to gateway 1 and prefix 99102 to gateway 2. Prefix have to be striped out before sending call to carrier. 2. Remote Call Forwarding. I have to setup RCF at Kamailio. All calls to the particular DID number have to be forward to another number. I found description for dialplan module and believe the module can work for me. I recompiled Kamailio with dialplan module support and added to config # ----- dialplan params ----- modparam("dialplan", "db_url", "mysql://openser:openserrw@localhost/openser") I did search and found this thread related to my problem: http://lists.sip-router.org/pipermail/sr-users/2009-November/025416.html I try to use kamctl to add rule to the MySQL and can't find proper syntax kamctl dialplan addrule 10 100 1 ^123+ 0 ^(123.+) 0030\1 -bash: syntax error near unexpected token `(' any help is greatly appreciated. Thank you.

3 3

Re: [SR-Users] module version mismatch for carrierroute
by Uri Shacked 24 Oct '11

24 Oct '11

Hi, i found out the mistake i made... didn't add the carrierroute to the include modules when making cfg file. thanks

1 0

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sr-users October 2011