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[Serusers] Problems with CANCEL and INVITE
by Benito Caracuel 08 Nov '06

08 Nov '06

Hello all, I have a problem when I try to cancel an invite, because I have modified the invite method. In my ser.cfg: if ((method=="INVITE")&&(!lookup("location"))) { exec_msg("printenv SRCIP; /bin/bash /usr/local/ser_snapshot/sbin/sms"); sl_send_reply("100", "Trying"); exec_msg("sleep 10"); if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("404", "Not on-line"); break; } else { t_relay(); break; } }; If I send an INVITE to user2, who is off-line, SER sends a SMS to this user2 and meanwhile , sends TRYINGS to me, until user2 sends a REGISTER. I have a problem, because If I send a CANCEL, SER still sends TRYINGS to me and SER doesn`t send "487 Request Terminated" and SER doesn´t associate CANCEL and INVITE. How can I solve this? any idea? I think that maybe, If I use a mysql table to save CSEQ of CANCEL, and before SER sends TRYINGS, SER checks if CSEQ is into table but I don´t know if SER can save CANCEL information in a mysql table. Could someone help me, please? Thank you very much. _________________________________________________________________ Prueba algunos de los nuevos servicios en línea que te ofrece Windows Live Ideas: tan nuevos que ni siquiera se han publicado oficialmente todavía. http://ideas.live.com

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[Serusers] Accounting mismatch
by inge 08 Nov '06

08 Nov '06

Hi all, When my users make a call, the domain field in "acc" table from SER database have a constant FQDN something like aa.mydomain.ext. But there is some case where the domain field contain IP of SER, not FQDN. The problem is that when I consult accounting page I only see the call where the domain field contains aa.mydomain.ext. Is it possible to see both call : FQDN/IP ? Thank for your help. Best regards.

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[Serusers] Cancelling an INVITE
by Benito Caracuel 08 Nov '06

08 Nov '06

Hello all, I have a problem when I try to cancel an invite, because I have modified the invite method. I'm going to explain the idea behind the problem, which is related to what we call "OffLine Invitation". An "Offline Invitation" consists of the SER set up as to DON'T send a "404 Not Found" when the Invitated party isn't registered. What is done is to keep the SER sending "100 Trying" for a while, and to send meanwhile a SMS to the Invitated -but not registered- party. That SMS is understood by the Application that's running at the Invited party as a "Register Request", so the Invitated party goes for a Register. That works OK, but the problem comes up when you try to CANCEL the Invitation request. In that case, as the Invitated party isn't still registered, it cannot take care of the CANCEL's lifecicle, so SER has to deal with it, "emulating" that it is the Invitated party. As to do this, we assume that no race condition is produced, and neither a Final response has been sent by the Invitated party. So what is needed is, according to SIP protocol, to answer to the CANCEL with "200 OK" and to the original INVITE with a "487 Request Terminated". The problem is obviously doing the INVITE - CANCEL match at the time the CANCEL gets to SER. We know there are scripts/procedures and tables in the SER architecture that can be used to do other tasks (as it is the "saving" of REGISTER packet onto a DB table), but, to be honest, we are a bit confused about if it can be done, or if it should need SER ad-hoc programming, or if there is a way to use an specific SER DB Table to do it... I post along my SER.CFG file, and an idea about how we were thinking that it could be solved (that is, a big picture; what we need indeed is to know if it is a good idea (having in mind SER architecture and scripting capabilities) or it cannot be done. Please, if it can be done, we'd like to hear a little more about what tables can be used, or if it's needed to create another new one, or at least, a clue to solve it. In my ser.cfg: if ((method=="INVITE")&&(!lookup("location"))) { exec_msg("printenv SRCIP; /bin/bash /usr/local/ser_snapshot/sbin/sms"); sl_send_reply("100", "Trying"); exec_msg("sleep 10"); if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("404", "Not on-line"); break; } else { t_relay(); break; } }; If I send an INVITE to user2, who is off-line (not registered), SER sends a SMS to this user2 and meanwhile, sends TRYINGS to me, until user2 becomes REGISTERED (the sent SMS makes the Invited party Register, so it can receive the Invite, as It was told before). I have a problem, because If I send a CANCEL, SER still sends TRYINGS to me and SER doesn`t send "487 Request Terminated" and SER doesn´t associate CANCEL and INVITE. How can I solve this? any idea? I think that maybe, If I use a mysql table to save CSEQ of CANCEL, and before SER sends TRYINGS (I mean the next TRYINGs following the first), SER checks if CSEQ is into table, but I don´t know if SER can save CANCEL information in a mysql table. I know that there is a command "save" to save REGISTER packet into location table. Is there any similar command to save CSEQ of CANCEL (and/or other, as Call-Id, Branch, Request-Uri and To) in one table? Could someone help me, please? We are in a hurry with our customer, so fast key to a solution will be really appreciated. Thank you very much. _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

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[Users] Not Hung Up when i Keep the Server in Router_firewall( NETGEAR)
by raviprakash sunkara 08 Nov '06

08 Nov '06

Hello User, A small Question , When I My Sip Server without any firewall_ Router .. Hung up is working Fine,,,,,, But I Keep Sip Server with in a Firewall_router, it not Hung upping , When i saw logs from X-lite in Bye method it Changing the uri into Private IP, But should it public IP( THis is not happening ) Please help me............. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Serusers] mediaproxy and ser ottendorf incompatible right now
by samuel 08 Nov '06

08 Nov '06

I was trying to use mediaproxy module with SER ottendorf and got next error: error: mediaproxy/mod_init(): can't find is_from_local and/or is_uri_host_local functions. Check if domain.so is loaded domain.so is effectively loaded BUT these functions have been replaced by is_local,lookup_domain, and get_did in new SER version Is AG Projects taking care of this issue?!? Can someone shed dome light on this issue!??!?!? Thanks a lot, Samuel.

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Re: [Users] TM : retransmission timers
by Rao Ramaratnamma 08 Nov '06

08 Nov '06

Greg, The email from you and others is giving me the type of most direct fair insights I am looking for . You know, just some practical "what is all this about" type of things for my selforientation. Just that openser link you are attaching unfortunately looks too much like openser marketing material. It is saying that ser was discontinued but actually the information I'm getting from others telling me that it has some features which are very important and openser do not have. With all respect, it is understandable that openser people write nicely about themselves and less nicely about others but it remains so confusing that I do not have confidence to move with this particular material. The openser info just appears too little credible for me. I think I am beginning to get the bottom line and I hope I will create my mind soon and if people have other information for me pls send it. Especially, objective technical information including this timer discussion is helping me. You know, better understanding what is the difference in "what they think is important". That would surely help me beyound the filosophical "cats or dogs, who is a better friend of man" information aqusition phase people surely like to be engaged in :-) Thank you all guys for keeping helping me a lot, rr ----- Original Message ---- From: Greg Fausak <lgfausak(a)gmail.com> To: Rao Ramaratnamma <raramarat(a)yahoo.com> Cc: Christian Schlatter <cs(a)unc.edu>; users(a)openser.org Sent: Wednesday, November 8, 2006 5:21:41 AM Subject: Re: [Users] TM : retransmission timers You can go to openser.org's website and read about howhow openser and ser are related. http://www.openser.org/index.php?option=com_content&task=view&id=40&Itemid=… They are now two separate bodies of code, maintained by two different groups. Each one has gone off and implemented and support what they think is important. Both projects are open source projects, anyone is certainly welcome to contribute to either. -g On Nov 7, 2006, at 4:15 PM, Rao Ramaratnamma wrote: the ser ottendorf announcement does mention improved timers. Cannot openser include this feature too and cannot I merge ser with openser for good timers? I am still trying to understand the difference between ser and openser but standart compliance seems to be very important matter! Cannot people provide me with some hints? I am sure that I am not the only who is asking the difference between ser and openser. ser documentation does not appear uptodate, but the software as sannounced appears impressive. I have already asked this question but did not receive any answer. thank you in advance! rr ----- Original Message ---- From: Christian Schlatter <cs(a)unc.edu> To: users(a)openser.org Sent: Tuesday, November 7, 2006 10:52:56 PM Subject: Re: [Users] TM : retransmission timers Greg Fausak wrote: > Hello, > > I believe this is a well known bug. > Granularity of timers is 1 second. So, if you sign up for a timer to > be fired in 1 second it will happen anywhere between 0 seconds and 1 > second. > 2 seconds will happen between 1 and 2 seconds. I usually set up my > timers to be 2, 2, 4, 8. There are VOIP providers that are pretty > sticky about > the first 500ms. If you are using one of them you're out of luck. Yes, there is a timer process that wakes up every second to perform retransmissions. I was actually quite surprised that OpenSER, which is known to be very standards compliant, does not follow the RFC 3261 retransmission timeouts. On the other hand, the RFC 3261 timeout values are just suggestions and standards compliant SIP UA must accept shorter timeouts. Still it would be nice if OpenSER would support sub second timers, this would allow for shorter fail-over times. Christian > > I believe SER has made timer changes to support more exact timer > intervals. They are a completely different camp, with a different feature > set (although they share the same roots). > > -g > > > On 11/7/06, Jean-François SMIGIELSKI <jf-smig(a)ibelgique.com> wrote: >> Hello, >> >> I made strange observations about the intervals between >> retransmissions with the TM module. >> In my experiments, I used the default parameters for the TM module >> timers, and I sent an INVITE that cannot receive answers (it has a >> well known R-URI pattern that is forwarded to a place and port that >> nobody listen). >> >> When reading RFC3261, I expected to see intervals between >> retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. >> >> But with OpenSER, (I have tested with the debian package 1.1.0-5 on a >> debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by >> myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s >> are spent (9 sendings). The first interval is sometomes very short >> (40ms). >> >> Altough I like the sequence of 4s separated transmissions, I do not >> know why the first interval is so short, and why there is no sending >> after 1s. >> >> Did anybody observed such behaviours? Are they normal? >> >> Thanks in advance! >> >> JF Smigielski. >> >> >> ________________________________________________________________________ >> iBELGIQUE, exprimez-vous ! >> http://web.ibelgique.com/ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] TM : retransmission timers
by Jean-François SMIGIELSKI 08 Nov '06

08 Nov '06

Hello, I made strange observations about the intervals between retransmissions with the TM module. In my experiments, I used the default parameters for the TM module timers, and I sent an INVITE that cannot receive answers (it has a well known R-URI pattern that is forwarded to a place and port that nobody listen). When reading RFC3261, I expected to see intervals between retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. But with OpenSER, (I have tested with the debian package 1.1.0-5 on a debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s are spent (9 sendings). The first interval is sometomes very short (40ms). Altough I like the sequence of 4s separated transmissions, I do not know why the first interval is so short, and why there is no sending after 1s. Did anybody observed such behaviours? Are they normal? Thanks in advance! JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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[Serusers] grandstream configuration -- help
by Ravi ***** 08 Nov '06

08 Nov '06

Hi users , I am using ser-0.96 , and every thing seems to work fine . but when I make calls from grandstream ATA .. its is working properly but the problem is when the other party hung the phone SER cannot send BYE packets to grandstream NAT ip . It is sending to grandstream private-ip which inturns delayed for acct.stop packet unless grandstream hung the phone x-lite, is O.K and its sending to SER as tel.no@Nat-ip-address but grandstream is sending to SER as tel.no@private-ip-address so in record-route SER is strictly following this and making some small mistake,, like sending Bye request to private-ip How to solve this : -) Thank You. Regards, Ravi.

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[Serusers] Problem with CANCEL messages and Cisco 7905
by Kristian Larsson 08 Nov '06

08 Nov '06

Hey! This is my first time posting to the list, although I'm a long time reader :) I have searched the archives as well as the internet and found no solution or answer to this so far. Perhaps my Google Kung Fu isn't what it should be. Anyhow, I have a cisco 7905 phone working together with a SER machine. When dialing to the 7905 phone from another phone and hanging up on the caller phone before answering the 7905, ie CANCEL before the call has actually been setup, I get a "SIP/2.0 481 Call Leg/Transaction Does Not Exist." back from the Cisco phone. It's a pretty basic configuration (basically the one used in the examples for call forwarding). What I want to know is how the phone identifies a Call Leg/Transaction. I have used ngrep quite extensively and found nothing. Most values in both the INVITE and CANCEL message are equal. I've also got a Thomson ATA box which does not show the same behaviour despite being connected to the same SER. What values are used to identify a transaction, how can I see if two messages are of the same transaction? Perhaps this is actually a user agent problem, but I don't really know where to turn for help with this 7905 (Cisco only provides support if you use their Call Manager solution). regards, Kristian. -- Kristian Larsson KLL-RIPE Network Engineer Net At Once [AS35706] email: kristian(a)netatonce.se irc: kll@ircnet phone: +46 470 592717 cell: +46 704 910401

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[Users] pstn-to-openser, 403 Forbiden
by Ion Minzu 08 Nov '06

08 Nov '06

Hello , I have connected openser with pstn through cisco. when I make a call from voip network to pstn it's ok. but from pstn to voip I have a problem:openser answers 403 forbiden. in openser I do the authorisation on mysql, I have disabled authorisation on sip gateway: if (src_ip!=X.X.X.X) { if (!www_authorize("DOMAIN.COM","subscriber")) { www_challenge("DOMAIN.COM","0"); exit; } }; What is the problem? X.X.X.X is cisco U X.X.X.X:54177 -> 172.17.6.2:5060 INVITE sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.1 7.6.2>..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Supported: timer,100rel..Min-SE: 1800..Cisco-Guid : 2422905184-1850675675-2419190351-3917593119..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBS CRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 6..Remote-Party-ID: <sip:022250699@X.X.X.X>;party=calling;screen=yes;privacy=off..Timestamp: 116 2983794..Contact: <sip:022250699@X.X.X.X:5060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 235....v= 0..o=CiscoSystemsSIP-GW-UserAgent 1226 5023 IN IP4 X.X.X.X..s=SIP Call..c=IN IP4 X.X.X.X..t=0 0..m=audio 16642 RTP/AVP 18 19..c=IN IP4 X.X.X.X..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:19 CN/8000..a=ptime:20.. # U 172.17.6.2:5060 -> X.X.X.X:5060 SIP/2.0 403 Use From=ID..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.17.6.2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..CSeq: 101 INVITE..Server: OpenSer (1.1.0-tls (x86_64/linux))..C ontent-Length: 0..Warning: 392 172.17.6.2:5060 "Noisy feedback tells: pid=32240 req_src_ip=X.X.X.X req_src_port=54177 in_uri=sip:820022@172.17.6.2:5 060 out_uri=sip:820022@172.17.6.2:5060 via_cnt==1".... # U X.X.X.X:54177 -> 172.17.6.2:5060 ACK sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.17.6 .2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Max-Forward s: 6..Content-Length: 0..CSeq: 101 ACK.... Best regards, Ion Minzu, Specialist Tehnologii Informationale, Administrator de sistem al Centrului de certificare, Administrator VoIP, I.S."Centrul de Telecomunicatii Speciale", tel:250-517 (office), 069501208 (mob), 382869185 (ICQ) mailto:ion.minzu@cts.md

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