- sr-users - kamailio.org

[OpenSER-Users] how to register xlite user for testing with openser
by Live Great 22 Sep '07

22 Sep '07

Hi, I used the following command to add xlite user, but xlite failed to register the user in openser server. # openserctl add 1234 1234 sam(a)my.domain.com where 1234 is user id and password. If this is not the correct way to add user for testing from xlite to openser, what are the alternative way I can use? Thanks SW

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[OpenSER-Users] OpenSER + FreeRadius error
by Marc LEURENT 22 Sep '07

22 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Does this error means something to somebody?? Thanks ERROR:acc:acc_rad_request: radius-ing failed My openser.cfg # -- acc params (with radius )-- modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) ..... ################### SET RADIUS ACCOUTING FLAGS ################################# # Set the acc flags xlog("JUSTE BEFORE SETTING FLAGS FOR RADIUS\r\n"); if(is_method("INVITE") && !has_totag()) { xlog("L_INFO", "I AM SETTING THE FLAGS FOR RADIUS \r\n"); $avp(can_uri) = $ru; # SIP Request's URI $avp(billing_party) = "sip:" + $au + "@" + $ar; # From URI $avp(user_name) = $fu; xlog("!!!! FROM TAG = $ft\r\n"); xlog("SETTING FLAGS 1 & 2 FOR RADIUS \r\n"); setflag(1); # radius_flag setflag(2); # radius_missed_flag }; DEBUG OUTPUT: 3(26026) DEBUG: update_totag_set: new totag 3(26026) DEBUG: add_to_tail_of_timer[2]: 0x2b898c7eb9d0 (12) 3(26026) DEBUG:tm:relay_reply: sent buf=0x61d2e8: SIP/2.0 2..., shmem=0x2b898c7ed7a8: SIP/2.0 2 3(26026) DBG: trans=0x2b898c7eb950, callback type 128, id 0 entered 3(26026) ERROR:acc:acc_rad_request: radius-ing failed 3(26026) DEBUG: cleanup_uac_timers: RETR/FR timers reset 3(26026) DEBUG:tm:UNREF_UNSAFE: after is 0 3(26026) DEBUG:destroy_avp_list: destroying list (nil) 3(26026) receive_msg: cleaning up 1(26024) SIP Request: 1(26024) method: <ACK> 1(26024) uri: <sip:105@82.127.0.79:1030;user=phone;nat=yes> 1(26024) version: <SIP/2.0> 1(26024) parse_headers: flags=2 1(26024) Found param type 232, <branch> = <z9hG4bK2175363103192047157>; state=16 1(26024) end of header reached, state=5 1(26024) parse_headers: Via found, flags=2 1(26024) parse_headers: this is the first via -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG8jcCqjpLE0HiOBYRAlBWAJoD4mlTLi8MCSLJEceI4jUNP2LFSwCgnjQq ytiBJX6Q40CVgt+V0sB0JWk= =BVYV -----END PGP SIGNATURE-----

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[Serusers] osp toolkit
by Arya 22 Sep '07

22 Sep '07

Hello Does osp toolkit support billing? and does it have a web interface? -- Thank You

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[OpenSER-Users] 1.3 release schedule
by Bogdan-Andrei Iancu 21 Sep '07

21 Sep '07

Hi, Just as reminder about the 1.3 release schedule: 05.10.2007 - SVN freeze and start testing - please commit all new features or changes till this date 0x.12.2007 - release date (first days of December) Regards, Bogdan

1 0

[OpenSER-Users] PA module
by Bogdan-Andrei Iancu 21 Sep '07

21 Sep '07

Hi, As we have in openser a new well designed presence support (see presence_* and pua_* modules), I think the PA module needs to be purged (anyhow it is not maintained). Does anybody has any good arguments not to do it? Regards, Bogdan

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[Serusers] Monitoring/logging SIP packets
by x-ser＠sidell.org 21 Sep '07

21 Sep '07

Sorry for the stupid question... Is there way to (temporarily) monitor and log the contents of all SIP packets being sent and received by my SER server? -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

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[OpenSER-Users] question about force_rtp_proxy()
by Papadopoulos Georgios 21 Sep '07

21 Sep '07

Hello all, I have come accross some issue while using nathelper and rtpproxy. The senario is as follows: An INVITE is sent to a NATed client and rtpproxy is used. force_rtp_proxy() rewrites the SDP to something like c=IN IP4 213.5.43.4. m=audio 35034 RTP/AVP 18 100 101. The client replies 486 Busy and the script sends the request to an Asterisk which is used for voicemail. If force_rtp_proxy() is called a second time before sending to Asterisk then the result is a SDP with the following. c=IN IP4 213.5.43.4213.5.43.4. m=audio 3503435034 RTP/AVP 18 100 101. Obviously this can be avoided if I add the proper logic in the script to avoid the second call to force_rtp_proxy(). However this behaviour looks like a bug. As a second question, when the call goes to Asterisk, there is no need for rtpproxy. So how can someone revert the SDP to its original form? How do you usually handle this case? thank you in advance George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[OpenSER-Users] can not acc failed transactions
by Victor Gamov 21 Sep '07

21 Sep '07

Hi All I red ACC module manual many times but still can not understand what I need to do to account failed transactions (408 for example) I make following configuration but failed transactions still not accounted ---------- cut here ------------------ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "db_flag", 1) route[1] { setflag(1); t_on_failure("1"); t_relay(); } failure_route[1] { xlog("L_ERR", "failure_route[1]: METHOD=$rm; rc=$T_reply_code\n"); } ---------- cut here ------------------ Yes I can add some acc_db_request() into failure_route[1] but this function don't set sip_code field in DB. Please advise me before I start to read sources :-) -- CU, Victor Gamov

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[Serusers] Cannot open module mysq.so - undefined symbol: log
by Borja Diaz de Teran 21 Sep '07

21 Sep '07

*Using Redhat Enterprise *MySQL Version 14.12 Distrib 5.0.27, for pc-linux-gnu (i686) using readline 5.0 1) I've followed the instructions to install SER in conjunction with Mysql and I've even tried to compile in a different way (make group_include="standard standard-dep stable mysql" skip_modules="postgres" all) [root@filipides ser]# ser -f /usr/local/etc/ser/ser.cfg Segment violation I keep getting the same error: Sep 21 13:57:30 filipides ser: ERROR: load_module: could not open module /usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Sep 21 13:57:30 filipides ser: parse error (81,13-14): failed to load module 2) LOG can be found in the following modules (apart from the ser exe):[root@filipides modules]# grep LOG * Match in cpl-c.so Match in dialog.so Match in mysql.so Match in pa.so Match in xlog.so 3) In order to check the existence of that pattern I did the following search:find . -name "*" -exec grep -l "LOG" {} \; , and it's inside (almost) all the files and objects compiled.implementación. The compilation of the MySQL module seems to be correct. gcc -shared -Wl,-O2 -Wl,-E dbase.o db_con.o db_mod.o my_con.o res.o row.o utils.o val.o -L/usr/lib/mysql -L/usr/local/lib -L/usr/local/lib/mysql -L/usr/local/mysql/lib/mysql/ -L/usr/local/mysql/lib -L/usr/lib64/mysql -lmysqlclient -lz -o mysql.so and it doesn't give any errors. I've tried changing all the LOG calls in all the .c files of the MySQL module to printf. I've compiled and install and nothing changes. I keep getting the same error.Do you have any clue of what I could be doing wrong?Thank you in advance. _________________________________________________________________ ¡Ya está aquí Windows Live Spaces! Ahora podrás crear fácilmente tu propio sitio Web. http://spaces.live.com/signup.aspx

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Re: [OpenSER-Users] Missing RTP stream
by Morten Isaksen 21 Sep '07

21 Sep '07

The error was in Mediaproxy. In rtphandler.py i changed nonPublicNetworks = [ {'name': '0.0.0.0', 'value': 0x00000000L, 'mask': 0xff000000L}, {'name': '10.0.0.0', 'value': 0x0a000000L, 'mask': 0xff000000L}, {'name': '127.0.0.0', 'value': 0x7f000000L, 'mask': 0xff000000L}, {'name': '172.16.0.0', 'value': 0xac100000L, 'mask': 0xfff00000L}, {'name': '192.168.0.0', 'value': 0xc0a80000L, 'mask': 0xffff0000L}, {'name': '224.0.0.0', 'value': 0xe0000000L, 'mask': 0xf0000000L} ] To nonPublicNetworks = [ {'name': '0.0.0.0', 'value': 0x00000000L, 'mask': 0xff000000L}, {'name': '10.0.0.0', 'value': 0x0a000000L, 'mask': 0xff000000L}, {'name': '127.0.0.0', 'value': 0x7f000000L, 'mask': 0xff000000L}, {'name': '224.0.0.0', 'value': 0xe0000000L, 'mask': 0xf0000000L} ] I think Mediaproxy got confused with the RFC1918 IP's. In my setup there is no NAT between 172.17.0.0/16 and 192.168.0.0/24 - just a router. On 9/21/07, Norman Brandinger <norm(a)goes.com> wrote: > Is there a firewall in the picture ? You have two different subnets and > there probably is a box doing some (NAT) translation / routing between > them. Is is possible the RTP stream is being blocked at the firewall ? > > Norm > > > Morten Isaksen wrote: > > Hi! > > > > I can see in the mediaproxy log the it is initialized to proxy the > > call, but I newer get a "session xxxxx: called signed in from xxx" > > from Asterisk. > > > > session.py shows that the the connection between mediaproxy and > > Asterisk is missing. > > > > I will try to take a look at the sip debug from asterisk and try to > > change the NAT settings in Asterisk. > > > > Thanks for your input. > > > > On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > > > >> You stated that you've forced every call through mediaproxy. Are you > >> positive ? > >> > >> Have you taken a look at the mediaproxy logs (and/or sessions.py when > >> the call is up) ? They might provide some useful information to you. > >> > >> Ditto for Asterisk "sip set debug on" (note that the sip debug command > >> format is a moving target). > >> > >> Have you looked at the "nat=" settings in sip.conf as well ? At times, > >> they tie closely with "canreinvite=". > >> > >> Norm > >> > >> > >> Morten Isaksen wrote: > >> > >>> Hi! > >>> > >>> canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the > >>> clients IP-addresses from Asterisk, so I am pretty sure that this is > >>> not the issue. > >>> > >>> On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > >>> > >>> > >>>> Hi Morten, > >>>> > >>>> Admittedly, I haven't looked closely at your trace. However, based on > >>>> the description you gave, the first place to look is at the "canrevite" > >>>> setting in Asterisk sip.conf. You might want to try "canreinvite=no" > >>>> after reading up on this particular setting. > >>>> > >>>> Regards, > >>>> Norm > >>>> > >>>> > >>>> Morten Isaksen wrote: > >>>> > >>>> > >>>>> Hi! > >>>>> > >>>>> I have a strange problem with a missing RTP stream between OpenSER and > >>>>> Asterisk. I am not sure if it is OpenSER og Asterisk related. > >>>>> > >>>>> I have this setup > >>>>> > >>>>> Phone A (172.17.96.17) -- > >>>>> \ Openser -- Asterisk > >>>>> -- PSTN > >>>>> / (192.168.0.6) (192.168.0.3) > >>>>> Phone B (172.17.96.10) -- (172.17.64.1) > >>>>> > >>>>> I also have a Mediaproxy running on OpenSER and I force every call to > >>>>> use the Mediaproxy. > >>>>> > >>>>> I call from Phone A or B to the PSTN works fine and from PSTN to Phone > >>>>> A or B it also works. > >>>>> > >>>>> I have the dialplan logic on my Asterisk server so I want calls from > >>>>> Phone A to Phone B to pass the Asterisk server. And this is were I > >>>>> have the problem. When the call is established the RTP stream is > >>>>> missing between Mediaproxy and Asterisk. I only have a RTP stream > >>>>> between the phones and Mediaproxy. As far as I can see the SIP > >>>>> signalling is correct. > >>>>> > >>>>> The SIP traces is listed below. Can you spot the problem in this? > >>>>> > >>>>> I will buy a beer (or 5) at OpenSER training in Rome to anyone who can > >>>>> help me solve this problem. > >>>>> > >>>>> SIP trace between the phones and OpenSER: > >>>>> > >>>>> > >>>>> > >>> > >>> > >> > > > > > > > > -- Morten Isaksen http://www.misak.dk/blog/

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[OpenSER-Users] Sending IM even when the destination User goes "Offline"
by Cesar Troncoso 21 Sep '07

21 Sep '07

Hello everybody, Lets suppose a user wants to send an Insant Message to another user of his contact list whose presence is "offline". Is there a way to store the message somewhere (another server maybe) for a certain period of time, so that when the destination user goes on-line again receives the Instant Message? Thank you very much in advanced for your help, Cesar

2 1

[OpenSER-Users] Authentication on SIP Invite Message
by Patrick Miccio 21 Sep '07

21 Sep '07

hi, you should have a look at the OpenSER modules like AUTH / AUTH_DB http://www.openser.org/docs/modules/ cheers, Patrick

2 1

[OpenSER-Users] Any way to cache SRV record lookups?
by Tim Madorma 21 Sep '07

21 Sep '07

Hi, I have an application where I am using the OpenSER with an SRV record lookup to load-balance towards several destination servers. When testing, I noticed that even if I have nscd caching turned on by setting: enable-cache hosts yes in my nscd.conf file, it still sends out a DNS SRV query for each call that is made (note that for A record lookups it does use the local nscd cache instead of doing a DNS query). I am running my host on Solaris. Is there any way to get nscd to cache SRV records? If not is there another alternative that I can use? thanks, Tim

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[OpenSER-Users] Authentication on SIP Invite Message
by Memon,Arshad Mehrab 21 Sep '07

21 Sep '07

Hi, Is there any way to enforce security on SIP invite Messages, The SIP proxy should send 407 (Proxy Authentication Required) in response to first INVITE. -- Best Regards, Arshad

1 0

[Serusers] about "calls_forwarding" table in SER database
by 作仁 21 Sep '07

21 Sep '07

Hi all Recently, I try to know how to use "calls_forwarding" table I had already search the document on the iptel.org website But, I can't find anything information about this table. Any one can tell me what value should stuff into this table? Thanks.

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[Serusers] New to SIP Express Server
by Do Nguyen Ha 21 Sep '07

21 Sep '07

hi all the SIP Express Server is a new thing to me Can anyone reccommend me where to start to understand & to config & to install the SIP Express Server + SEMS Thanks Do Nguyen Ha

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Re: [OpenSER-Users] creating call number for xlite user
by Live Great 21 Sep '07

21 Sep '07

Thanks for the info. In the openser server, can you tell me how to create phone number that allow xlite user to dial in (for testing purpose)? Thanks S ----- Original Message ---- From: Norman Brandinger <norm(a)goes.com> To: Live Great <livegreat007(a)yahoo.com> Cc: users(a)openser.org Sent: Friday, September 21, 2007 11:16:31 AM Subject: Re: [OpenSER-Users] creating call number for xlite user Executing openserctl without any parameters will display some usage information. You probably want to start out adding a user to the subscriber table. They syntax is shown below: openserctl add <username> <password> <email> rpid is the Remote-Party-ID and is a SIP header. This is defined in the following EXPIRED draft: http://tools.ietf.org/html/draft-ietf-sip-privacy-04 There are others on this list that might know what the current status of this header is. You may also want to research the P-Asserted-Identity (RFC3325) SIP header. There is a lot of documentation on the openser.org web site. Regards, Norm Live Great wrote: > Hi, > > I have completed the configuration of openser in freebsd 6.2 and > downloaded xlite to my windows xp. > In the man page of opernserctl, I am not fully understand how to add a > calling number for remote xlite user. > eg. what is rpid? the man page doesn't explain. > > Can anyone help? > > Thanks > Sam > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] creating call number for xlite user
by Live Great 21 Sep '07

21 Sep '07

Hi, I have completed the configuration of openser in freebsd 6.2 and downloaded xlite to my windows xp. In the man page of opernserctl, I am not fully understand how to add a calling number for remote xlite user. eg. what is rpid? the man page doesn't explain. Can anyone help? Thanks Sam

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[OpenSER-Users] Asterisk + OpenSER + WeSIP + Cisco Call Manager
by Victor Pascual Ávila 20 Sep '07

20 Sep '07

Hello, we're deploying a new ToIP network and we are currently using Asterisk (for some services), and OpenSER (as a trunk). Now we would like to introduce a Cisco Call Manager (registered to the OpenSER) and a WeSIP. Do you have any experience connecting CCM (using SIP) to OpenSER? What about WeSIP interaction with other different proxies (not OpenSER+SEAS)? Thanks in advance, Victor Pascual

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Re: [OpenSER-Users] Missing RTP stream
by Morten Isaksen 20 Sep '07

20 Sep '07

Hi! I can see in the mediaproxy log the it is initialized to proxy the call, but I newer get a "session xxxxx: called signed in from xxx" from Asterisk. session.py shows that the the connection between mediaproxy and Asterisk is missing. I will try to take a look at the sip debug from asterisk and try to change the NAT settings in Asterisk. Thanks for your input. On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > You stated that you've forced every call through mediaproxy. Are you > positive ? > > Have you taken a look at the mediaproxy logs (and/or sessions.py when > the call is up) ? They might provide some useful information to you. > > Ditto for Asterisk "sip set debug on" (note that the sip debug command > format is a moving target). > > Have you looked at the "nat=" settings in sip.conf as well ? At times, > they tie closely with "canreinvite=". > > Norm > > > Morten Isaksen wrote: > > Hi! > > > > canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the > > clients IP-addresses from Asterisk, so I am pretty sure that this is > > not the issue. > > > > On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > > > >> Hi Morten, > >> > >> Admittedly, I haven't looked closely at your trace. However, based on > >> the description you gave, the first place to look is at the "canrevite" > >> setting in Asterisk sip.conf. You might want to try "canreinvite=no" > >> after reading up on this particular setting. > >> > >> Regards, > >> Norm > >> > >> > >> Morten Isaksen wrote: > >> > >>> Hi! > >>> > >>> I have a strange problem with a missing RTP stream between OpenSER and > >>> Asterisk. I am not sure if it is OpenSER og Asterisk related. > >>> > >>> I have this setup > >>> > >>> Phone A (172.17.96.17) -- > >>> \ Openser -- Asterisk > >>> -- PSTN > >>> / (192.168.0.6) (192.168.0.3) > >>> Phone B (172.17.96.10) -- (172.17.64.1) > >>> > >>> I also have a Mediaproxy running on OpenSER and I force every call to > >>> use the Mediaproxy. > >>> > >>> I call from Phone A or B to the PSTN works fine and from PSTN to Phone > >>> A or B it also works. > >>> > >>> I have the dialplan logic on my Asterisk server so I want calls from > >>> Phone A to Phone B to pass the Asterisk server. And this is were I > >>> have the problem. When the call is established the RTP stream is > >>> missing between Mediaproxy and Asterisk. I only have a RTP stream > >>> between the phones and Mediaproxy. As far as I can see the SIP > >>> signalling is correct. > >>> > >>> The SIP traces is listed below. Can you spot the problem in this? > >>> > >>> I will buy a beer (or 5) at OpenSER training in Rome to anyone who can > >>> help me solve this problem. > >>> > >>> SIP trace between the phones and OpenSER: > >>> > >>> > > > > > > > > -- Morten Isaksen http://www.misak.dk/blog/

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[OpenSER-Users] Missing RTP stream
by Morten Isaksen 20 Sep '07

20 Sep '07

Hi! I have a strange problem with a missing RTP stream between OpenSER and Asterisk. I am not sure if it is OpenSER og Asterisk related. I have this setup Phone A (172.17.96.17) -- \ Openser -- Asterisk -- PSTN / (192.168.0.6) (192.168.0.3) Phone B (172.17.96.10) -- (172.17.64.1) I also have a Mediaproxy running on OpenSER and I force every call to use the Mediaproxy. I call from Phone A or B to the PSTN works fine and from PSTN to Phone A or B it also works. I have the dialplan logic on my Asterisk server so I want calls from Phone A to Phone B to pass the Asterisk server. And this is were I have the problem. When the call is established the RTP stream is missing between Mediaproxy and Asterisk. I only have a RTP stream between the phones and Mediaproxy. As far as I can see the SIP signalling is correct. The SIP traces is listed below. Can you spot the problem in this? I will buy a beer (or 5) at OpenSER training in Rome to anyone who can help me solve this problem. SIP trace between the phones and OpenSER: U 172.17.96.17:5060 -> 172.17.64.1:5060 INVITE sip:43336000@172.17.64.1:5060;transport=UDP SIP/2.0. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 1 INVITE. Via: SIP/2.0/UDP 172.17.96.17:5060;rport;branch=z9hG4bK-3e125d63-77bccac7-4b008a75. Max-Forwards: 70. Supported: replaces,100rel. Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 366. . v=0. o=081701 1041391881 1041391881 IN IP4 172.17.96.17. s=-. c=IN IP4 172.17.96.17. t=0 0. m=audio 10000 RTP/AVP 8 8 18 4 98 2 101. a=rtpmap:8 PCMA/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 g723/8000. a=rtpmap:98 g726-16/8000. a=rtpmap:2 g726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:18 annexb=no. a=fmtp:4 annexa=no. a=fmtp:101 0-15. # U 172.17.64.1:5060 -> 172.17.96.17:5060 SIP/2.0 407 Proxy Authentication Required. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=e7a9ec63e83496804ae077e38a2ef69c.8352. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 1 INVITE. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccac7-4b008a75. Proxy-Authenticate: Digest realm="172.17.96.17", nonce="46f2861ed3a501550cd33c28b12936d24c42199c". Server: OpenSer (1.1.0-notls (i386/linux)). Content-Length: 0. Warning: 392 172.17.64.1:5060 "Noisy feedback tells: pid=15649 req_src_ip=172.17.96.17 req_src_port=5060 in_uri=sip:43336000@172.17.64.1:5060;transport=UDP out_uri=sip:43336000@172.17.64.1:5060;transport=UDP via_cnt==1". . # U 172.17.96.17:5060 -> 172.17.64.1:5060 ACK sip:43336000@172.17.64.1:5060;transport=UDP SIP/2.0. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=e7a9ec63e83496804ae077e38a2ef69c.8352. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 1 ACK. Via: SIP/2.0/UDP 172.17.96.17:5060;rport;branch=z9hG4bK-3e125d63-77bccac7-4b008a75. Max-Forwards: 70. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Content-Length: 0. . # U 172.17.96.17:5060 -> 172.17.64.1:5060 INVITE sip:43336000@172.17.64.1:5060;transport=UDP SIP/2.0. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. Via: SIP/2.0/UDP 172.17.96.17:5060;rport;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. Max-Forwards: 70. Supported: replaces,100rel. Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Proxy-Authorization: Digest username="081701",realm="172.17.96.17",nonce="46f2861ed3a501550cd33c28b12936d24c42199c",uri="sip:43336000@172.17.64.1:5060;transport=UDP",response="4bf0194044fb4a759ec6134d68bef752",algorithm=MD5. Content-Type: application/sdp. Content-Length: 366. . v=0. o=081701 1041391881 1041391881 IN IP4 172.17.96.17. s=-. c=IN IP4 172.17.96.17. t=0 0. m=audio 10000 RTP/AVP 8 8 18 4 98 2 101. a=rtpmap:8 PCMA/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 g723/8000. a=rtpmap:98 g726-16/8000. a=rtpmap:2 g726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:18 annexb=no. a=fmtp:4 annexa=no. a=fmtp:101 0-15. # U 172.17.64.1:5060 -> 172.17.96.17:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3:5060>. Content-Length: 0. . # U 172.17.64.1:5060 -> 172.17.96.10:5060 INVITE sip:081700@172.17.96.10:5060;transport=UDP SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=as28ccd5e9>. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.1. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>. Contact: <sip:43336999@192.168.0.3:1265>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 69. Date: Thu, 20 Sep 2007 14:34:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 182. . v=0. o=root 21472 21472 IN IP4 192.168.0.3. s=session. c=IN IP4 172.17.64.1. t=0 0. m=audio 60024 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U 172.17.64.1:5060 -> 172.17.96.17:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3:5060>. Content-Length: 0. . # U 172.17.96.10:5060 -> 172.17.64.1:5060 SIP/2.0 180 Ringing. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.1,SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Content-Length: 0. . # U 172.17.96.10:5060 -> 172.17.64.1:5060 SIP/2.0 200 OK. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.1,SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Record-Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. Contact: <sip:081700@172.17.96.10:5060;transport=UDP>. Content-Length: 171. Content-Type: application/sdp. . v=0. o=- 926679040 1 IN IP4 172.17.96.10. s=ATI-SIP Call. c=IN IP4 172.17.96.10. t=0 0. m=audio 50600 RTP/AVP 8 13. a=rtpmap:8 PCMA/8000. a=ptime:10. a=rtpmap:13 CN/8000. # U 172.17.64.1:5060 -> 172.17.96.12:5060 CANCEL sip:081700@172.17.96.12:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.0. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. To: <sip:081700@172.17.64.1>. CSeq: 102 CANCEL. User-Agent: OpenSer (1.1.0-notls (i386/linux)). Content-Length: 0. . # U 172.17.64.1:5060 -> 172.17.96.10:5060 ACK sip:081700@172.17.96.10:5060;transport=UDP SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=as28ccd5e9>. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.3. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK78618ac6;rport=1265. Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Contact: <sip:43336999@192.168.0.3>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 69. Content-Length: 0. . # U 172.17.64.1:5060 -> 172.17.96.17:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. Record-Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3:5060>. Content-Type: application/sdp. Content-Length: 238. . v=0. o=root 21472 21472 IN IP4 192.168.0.3. s=session. c=IN IP4 172.17.64.1. t=0 0. m=audio 60022 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 172.17.96.17:5060 -> 172.17.64.1:5060 ACK sip:43336000@192.168.0.3:5060 SIP/2.0. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 ACK. Via: SIP/2.0/UDP 172.17.96.17:5060;rport;branch=z9hG4bK-3e125d66-77bcd96f-26ad0a63. Max-Forwards: 70. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Route: <sip:172.17.64.1;lr;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. Proxy-Authorization: Digest username="081701",realm="172.17.96.17",nonce="46f2861ed3a501550cd33c28b12936d24c42199c",uri="sip:43336000@172.17.64.1:5060;transport=UDP",response="4bf0194044fb4a759ec6134d68bef752",algorithm=MD5. Content-Length: 0. . # U 172.17.96.10:5060 -> 172.17.64.1:5060 BYE sip:43336999@192.168.0.3:1265 SIP/2.0. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. Max-Forwards: 70. Supported: 100rel,replaces,timer. Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. Content-Length: 0. . # U 172.17.64.1:5060 -> 172.17.96.10:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336999@192.168.0.3:5060>. Content-Length: 0. X-Asterisk-HangupCause: Normal Clearing. . # U 172.17.64.1:5060 -> 172.17.96.10:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336999@192.168.0.3:1265>. Content-Length: 0. X-Asterisk-HangupCause: Normal Clearing. . # U 172.17.64.1:5060 -> 172.17.96.17:5060 BYE sip:081701@172.17.96.17:5060 SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=as119e7abe>. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bK4aff.f48c6cd7.0. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0716a581;rport=5060. Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. To: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. Contact: <sip:43336000@192.168.0.3>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 102 BYE. User-Agent: Asterisk PBX. Max-Forwards: 69. Content-Length: 0. . # U 172.17.96.17:5060 -> 172.17.64.1:5060 SIP/2.0 200 OK. From: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. To: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 102 BYE. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bK4aff.f48c6cd7.0. Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;branch=z9hG4bK0716a581. Supported: replaces,100rel. Content-Length: 0. . And the SIP trace between OpenSER and Asterisk: # U 192.168.0.6:5060 -> 192.168.0.3:5060 INVITE sip:43336000@172.17.64.1:5060;transport=UDP SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. Via: SIP/2.0/UDP 172.17.64.1;branch=0. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. Max-Forwards: 69. Supported: replaces,100rel. Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Proxy-Authorization: Digest username="081701",realm="172.17.96.17",nonce="46f2861ed3a501550cd33c28b12936d24c42199c",uri="sip:43336000@172.17.64.1:5060;transport=UDP",response="4bf0194044fb4a759ec6134d68bef752",algorithm=MD5. Content-Type: application/sdp. Content-Length: 365. . v=0. o=081701 1041391881 1041391881 IN IP4 172.17.96.17. s=-. c=IN IP4 172.17.64.1. t=0 0. m=audio 60022 RTP/AVP 8 8 18 4 98 2 101. a=rtpmap:8 PCMA/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 g723/8000. a=rtpmap:98 g726-16/8000. a=rtpmap:2 g726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:18 annexb=no. a=fmtp:4 annexa=no. a=fmtp:101 0-15. # U 192.168.0.3:5060 -> 192.168.0.6:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.17.64.1;branch=0;received=192.168.0.6. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3>. Content-Length: 0. . # U 192.168.0.3:5060 -> 172.17.64.1:5060 INVITE sip:081700@172.17.64.1 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>. Contact: <sip:43336999@192.168.0.3>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Thu, 20 Sep 2007 14:34:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 182. . v=0. o=root 21472 21472 IN IP4 192.168.0.3. s=session. c=IN IP4 192.168.0.3. t=0 0. m=audio 10810 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U 172.17.64.1:5060 -> 192.168.0.3:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Server: OpenSer (1.1.0-notls (i386/linux)). Content-Length: 0. Warning: 392 172.17.64.1:5060 "Noisy feedback tells: pid=15647 req_src_ip=192.168.0.3 req_src_port=1265 in_uri=sip:081700@172.17.64.1 out_uri=sip:081700@172.17.96.12:5060;transport=UDP via_cnt==1". . # U 192.168.0.3:5060 -> 192.168.0.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.17.64.1;branch=0;received=192.168.0.6. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3>. Content-Length: 0. . # U 172.17.64.1:5060 -> 192.168.0.3:5060 SIP/2.0 180 Ringing. User-Agent: ATI-RG613TX/2-3_59. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAAGNwAN2gAZewxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Content-Length: 0. . # U 172.17.64.1:5060 -> 192.168.0.3:5060 SIP/2.0 180 Ringing. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Content-Length: 0. . # U 172.17.64.1:5060 -> 192.168.0.3:5060 SIP/2.0 200 OK. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Record-Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. Contact: <sip:081700@172.17.96.10:5060;transport=UDP>. Content-Length: 170. Content-Type: application/sdp. . v=0. o=- 926679040 1 IN IP4 172.17.96.10. s=ATI-SIP Call. c=IN IP4 172.17.64.1. t=0 0. m=audio 60024 RTP/AVP 8 13. a=rtpmap:8 PCMA/8000. a=ptime:10. a=rtpmap:13 CN/8000. # U 192.168.0.3:5060 -> 172.17.64.1:5060 ACK sip:081700@172.17.96.10:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK78618ac6;rport. Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Contact: <sip:43336999@192.168.0.3>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U 192.168.0.3:5060 -> 192.168.0.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.64.1;branch=0;received=192.168.0.6. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. Record-Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3>. Content-Type: application/sdp. Content-Length: 238. . v=0. o=root 21472 21472 IN IP4 192.168.0.3. s=session. c=IN IP4 192.168.0.3. t=0 0. m=audio 12684 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 192.168.0.6:5060 -> 192.168.0.3:5060 ACK sip:43336000@192.168.0.3:5060 SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 ACK. Via: SIP/2.0/UDP 172.17.64.1;branch=0. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d66-77bcd96f-26ad0a63. Max-Forwards: 69. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Route: <sip:172.17.64.1;lr;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. Proxy-Authorization: Digest username="081701",realm="172.17.96.17",nonce="46f2861ed3a501550cd33c28b12936d24c42199c",uri="sip:43336000@172.17.64.1:5060;transport=UDP",response="4bf0194044fb4a759ec6134d68bef752",algorithm=MD5. Content-Length: 0. . # U 192.168.0.6:5060 -> 192.168.0.3:5060 BYE sip:43336999@192.168.0.3:1265 SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.64.1;branch=0. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. Max-Forwards: 69. Supported: 100rel,replaces,timer. Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. Content-Length: 0. . # U 172.17.64.1:5060 -> 192.168.0.3:5060 BYE sip:43336999@192.168.0.3:1265 SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bK26ae.adc1e03.0. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. Max-Forwards: 69. Supported: 100rel,replaces,timer. Content-Length: 0. P-hint: rr-enforced. . # U 192.168.0.3:5060 -> 192.168.0.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.64.1;branch=0;received=192.168.0.6. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336999@192.168.0.3>. Content-Length: 0. X-Asterisk-HangupCause: Normal Clearing. . # U 192.168.0.3:5060 -> 172.17.64.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bK26ae.adc1e03.0;received=172.17.64.1. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336999@192.168.0.3>. Content-Length: 0. X-Asterisk-HangupCause: Normal Clearing. . # U 192.168.0.3:5060 -> 192.168.0.6:5060 BYE sip:081701@172.17.96.17:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0716a581;rport. Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. To: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. Contact: <sip:43336000@192.168.0.3>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 102 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U 192.168.0.6:5060 -> 192.168.0.3:5060 SIP/2.0 200 OK. From: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. To: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 102 BYE. Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;branch=z9hG4bK0716a581. Supported: replaces,100rel. Content-Length: 0. . -- Morten Isaksen http://www.misak.dk/blog/

2 2

[OpenSER-Users] Forward REGISTER to Asterisk
by Halomoan Chow 20 Sep '07

20 Sep '07

Dear All I want user who register with my openser also register with my asterisk for later lookup. I tried t_replicate function from TM module but no luck. I end up with Unauthorized message. Could any help me how to make this work? What do I miss ? Thank so much Regards, Halomoan Zhou

4 4

Re: [OpenSER-Users] test operator]
by Klaus Darilion 20 Sep '07

20 Sep '07

Bogdan-Andrei Iancu schrieb: > Hi Klaus, > > the [..] operator is used to convert generic numerical value (maybe > result of arithmetic ops to logical value. > > in your example, the == is already a logical operator, so no need to any > conversion. Does ($rd == "mydomain") works too? is this somewhere documented? The corebook only describes assignments, but not conditions/comparisons. regards klaus

2 1

[OpenSER-Users] tcp receiver with tcp socket?
by Klaus Darilion 20 Sep '07

20 Sep '07

Hi! I have configured openser 1.2 to only listen to udp: Starting Open SIP Router: Listening on udp: 1.2.3.4 [1.2.3.4]:5060 Aliases: *: mydomain.at:* $ openser-1.2-SVN/scripts/openserctl ps Process:: ID=0 PID=29582 Type=attendant Process:: ID=1 PID=29583 Type=receiver child=0 sock= 1.2.3.4:5060 Process:: ID=2 PID=29585 Type=receiver child=1 sock= 1.2.3.4:5060 Process:: ID=3 PID=29586 Type=receiver child=2 sock= 1.2.3.4:5060 Process:: ID=4 PID=29587 Type=receiver child=3 sock= 1.2.3.4:5060 Process:: ID=5 PID=29588 Type=receiver child=4 sock= 1.2.3.4:5060 Process:: ID=6 PID=29589 Type=receiver child=5 sock= 1.2.3.4:5060 Process:: ID=7 PID=29590 Type=receiver child=6 sock= 1.2.3.4:5060 Process:: ID=8 PID=29591 Type=receiver child=7 sock= 1.2.3.4:5060 Process:: ID=9 PID=29592 Type=timer Process:: ID=10 PID=29593 Type=tcp receiver Process:: ID=11 PID=29594 Type=tcp receiver Process:: ID=12 PID=29595 Type=tcp receiver Process:: ID=13 PID=29596 Type=tcp receiver Process:: ID=14 PID=29597 Type=tcp receiver Process:: ID=15 PID=29598 Type=tcp receiver Process:: ID=16 PID=29599 Type=tcp receiver Process:: ID=17 PID=29600 Type=tcp receiver Process:: ID=18 PID=29601 Type=tcp main process So, where does all the "tcp receiver" come from? If they are DB related maybe we should change the "Type". This proxy uses mysql for presence - this I guess all the TCP connections are for presence. But why do I have 19 mysql connections? $ netstat -anp|grep openser tcp 0 0 1.2.3.4:48896 1.2.3.68:3306 ESTABLISHED 29598/openser tcp 0 0 1.2.3.4:48897 1.2.3.68:3306 ESTABLISHED 29599/openser tcp 0 0 1.2.3.4:48898 1.2.3.68:3306 ESTABLISHED 29600/openser tcp 0 0 1.2.3.4:48899 1.2.3.68:3306 ESTABLISHED 29601/openser tcp 0 0 1.2.3.4:48900 1.2.3.68:3306 ESTABLISHED 29582/openser tcp 0 0 1.2.3.4:48889 1.2.3.68:3306 ESTABLISHED 29591/openser tcp 0 0 1.2.3.4:48888 1.2.3.68:3306 ESTABLISHED 29590/openser tcp 0 0 1.2.3.4:48891 1.2.3.68:3306 ESTABLISHED 29593/openser tcp 0 0 1.2.3.4:48890 1.2.3.68:3306 ESTABLISHED 29592/openser tcp 0 0 1.2.3.4:48893 1.2.3.68:3306 ESTABLISHED 29595/openser tcp 0 0 1.2.3.4:48892 1.2.3.68:3306 ESTABLISHED 29594/openser tcp 0 0 1.2.3.4:48895 1.2.3.68:3306 ESTABLISHED 29597/openser tcp 0 0 1.2.3.4:48894 1.2.3.68:3306 ESTABLISHED 29596/openser tcp 0 0 1.2.3.4:48883 1.2.3.68:3306 ESTABLISHED 29585/openser tcp 0 0 1.2.3.4:48882 1.2.3.68:3306 ESTABLISHED 29583/openser tcp 0 0 1.2.3.4:48885 1.2.3.68:3306 ESTABLISHED 29587/openser tcp 0 0 1.2.3.4:48884 1.2.3.68:3306 ESTABLISHED 29586/openser tcp 0 0 1.2.3.4:48887 1.2.3.68:3306 ESTABLISHED 29589/openser tcp 0 0 1.2.3.4:48886 1.2.3.68:3306 ESTABLISHED 29588/openser udp 0 0 1.2.3.4:5060 0.0.0.0:* 29582/openser Thus, why 19 DB connections for 8 UDP childrens? Why 8 tcp receivers in "openserctl ps"? thanks klaus

2 2

[OpenSER-Users] test operator
by Klaus Darilion 20 Sep '07

20 Sep '07

Hi! from the core cookbook of 1.2: > If you want to evaluate arithmetic operations and test the value in > conditions, use the test operator ' [ ... ] ' . This is required to > convert the result to internal convention for drop, true and false. > > if ( [ $var(a)&4 ] ) { > xlog("bit 3 is set in var a"); > } Is there somewhere a detailed description of the test operator? Can it only be used to test for 0/1 or can I also do comparison like e.g. if ([ $rd == "mydomain.at" ]) ? regards klaus

2 1

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