sr-users April 2007

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[Users] use of AVPs in core functions
by Jens Carl 02 Apr '07

02 Apr '07

Hello List, is it possible to use AVPs in core function like rewritehost(), etc. Regards Jens

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Re: [Users] Openser on Sun Ultrasparc T1 Processors - Mismatch architecture detection
by Henning Westerholt 02 Apr '07

02 Apr '07

On Tuesday 27 March 2007 17:41, you wrote: > Hi Henning. > > I tested the compilation and it does not detect the architecture. > > I think that maybe is because of uname -p outputs sparc, and uname -m > outputs sun4v. > [..] Hi Sergio, thank you for the testing. I add another check to the patch according your suggestions, could please try it another time? Cheers, Henning

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re: [Users] postgres
by Mark Price 02 Apr '07

02 Apr '07

On 12/20/06, Klaus Darilion <klaus.mailinglists(a)pernau.at > wrote: > > Hi Mark! > > Postgres should work well - I use it since ser 0.8. Just make sure that > the hard disk does not get full, because this breaks the index inside > postgres and postgres is getting real slow (re-create the index if it > happens). > > Last time I tested openser_postgres.sh it worked fine. If you find a bug > please let us know. > > Please check to use the latest versions from CVS (for 1.1 use CVS > rel_1_1_0) The latest version from cvs doesn't work out of the box with openser build from the release tarball, because the release tarball doesn't included /usr/sbin/openser_gen_ha1 (although I could get past this by replacing it with the md5sum equivelant). However, the release tarball doesn't work because of this bug: https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_… So, 1. is the cvs version of rel_1_1_0 considered safe for production use? 2. if so and there are important fixes such as this in CVS, would openser.org consider releasing a point release containing such changes? Thanks, Mark Price

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RE: [Users] out of memory
by Papadopoulos Georgios 02 Apr '07

02 Apr '07

Hi Bogdan, I increased PKG_MEM_POOL_SIZE to 2MB and of course it took much longer to run out of memory. Usually it happens once every day, this time it ran for 3 days before running out of memory. If you don't mind, I will send the memory dump by personal email since I don't have easy access to an ftp server. Can you explain what you mean by 1.1.x? The version I am using is openser-1.1.1-notls_src.tar.gz, downloaded from your site under 1.1.1/. I am guessing that to get the fix I should use whatever is under latest-1.1.x/ Is this right? thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Friday, March 30, 2007 3:33 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] out of memory > > Papadopoulos Georgios wrote: > > > > Hi Bogdan, > > > > I found the following thread in the list > > http://www.openser.org/pipermail/users/2006-June/005339.html > > I suspect that I have the same issue. If I assume that our DNS is > > strange, then my 1.1.1-notls version will be affected, > right? Is there > > a patch to fix this? Eventually I want to go to 1.2 but I > have to do a > > lot of testing before I can put it in production. So for > the moment I > > need a fix for 1.1.1. > > > that fix is present in 1.1.x also. > > Also back in December I sent this to the list > > http://www.openser.org/pipermail/users/2006-December/008205.html > > >From the start, we had the issue of openser stopping to process any > > incoming requests. On December I thought it was DNS related. I am > > still in the dark about this. Lately I added logging and I > discovered > > the "out of memory" error. I am guessing that the error was > there from > > the start, I just did not know about it. Do all of these make any > > sense to you? And is there a way to test our DNS to > determine whether it is buggy? > > > I;m afraid not :) > > In the meantime I will follow the instructions on the wiki > to get the > > memory dump. > > > yes - that will be the most relevant info. > > regards, > bogdan > > thank you > > > > George > > > > > > > > > >> -----Original Message----- > >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >> Sent: Wednesday, March 28, 2007 11:54 AM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] out of memory > >> > >> Hi Papadopoulos, > >> > >> the error you get complains about the lack of private memory (per > >> process) - the statistics show info about the shared memory. See: > >> http://www.openser.org/dokuwiki/doku.php/troubleshooting:memory > >> > >> but look for pkg instead of shm..... > >> > >> regards, > >> bogdan > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hello, > >>> > >>> Lately our OpenSER 1.1.1 is running out of memory and I get the > >>> following message > >>> 4(25190) get_hdr_field: out of memory The result is that > it stops > >>> processing any incoming request. This is a production > system and of > >>> course it causes a great deal of frustration. > >>> > >>> The last time this happened I got the following statistics before > >>> restarting > >>> > >>>> ./openserctl fifo get_statistics > >>>> > >>> Total statistics = 37 > >>> Total modules = 5 > >>> Module name = core; statistics=8 > >>> core:rcv_requests = 3259334 > >>> core:rcv_replies = 8816 > >>> core:fwd_requests = 736 > >>> core:fwd_replies = 137 > >>> core:drop_requests = 743 > >>> core:drop_replies = 0 > >>> core:err_requests = 0 > >>> core:err_replies = 43 > >>> Module name = shmem; statistics=6 > >>> shmem:total_size = 268435456 > >>> shmem:used_size = 1675864 > >>> shmem:real_used_size = 1835104 > >>> shmem:max_used_size = 4399048 > >>> shmem:free_size = 266600352 > >>> shmem:fragments = 17710 > >>> Module name = sl; statistics=9 > >>> sl:1xx_replies = 0 > >>> sl:2xx_replies = 1690410 > >>> sl:3xx_replies = 43 > >>> sl:4xx_replies = 334827 > >>> sl:5xx_replies = 1185629 > >>> sl:6xx_replies = 0 > >>> sl:sent_replies = 3210909 > >>> sl:sent_err_replies = 0 > >>> sl:received_ACKs = 743 > >>> Module name = tm; statistics=11 > >>> tm:received_replies = 8636 > >>> tm:relayed_replies = 7096 > >>> tm:local_replies = 1718 > >>> tm:UAS_transactions = 7003 > >>> tm:UAC_transactions = 94 > >>> tm:2xx_transactions = 1528 > >>> tm:3xx_transactions = 3 > >>> tm:4xx_transactions = 5129 > >>> tm:5xx_transactions = 468 > >>> tm:6xx_transactions = 0 > >>> tm:inuse_transactions = 0 > >>> Module name = usrloc; statistics=3 > >>> usrloc:location-users = 0 > >>> usrloc:location-contacts = 0 > >>> usrloc:location-expires = 7282 > >>> > >>> I am not sure how to read these numbers but I see a big > "free_size" > >>> and still OpenSER is out of memory. Any suggestions? > >>> > >>> thank you > >>> > >>> 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. > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >> - > >> > >>> -- > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >> > > > > > >

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[Users] OpenSER v1.2.0 released
by Daniel-Constantin Mierla 02 Apr '07

02 Apr '07

Hello, version 1.2.0 of OpenSER has been released. It is major release, which means that you get along a lot of new features and improvements to existing ones in old versions. The configuration file and database structure suffered some changers, please read careful release notes to learn how to update. Just to list few of the new features: - SIMPLE Presence support (http://openser.org/dokuwiki/doku.php/presence:presence-module) - SNMP (http://openser.org/docs/modules/devel/snmpstats.html) - PERL API (http://openser.org/docs/modules/devel/perl.html) - JAVA SIP Servlet Application Server (http://wesip.eu) - script variables and transformations (http://openser.org/dokuwiki/doku.php/transformations:devel) - DNS failover - IP blacklists - XMPP IM gateway - XMLRPC - SIP session timers - secure federation peering - retransmission timer accuracy - usrloc refurbished for better performances - automatic error handling - AVPs in reply routes Read full release notes at: http://www.openser.org/mos/view/OpenSER-v1.2.x-Release-Notes/ Source tarball can be downloaded from: http://www.openser.org/pub/openser/latest/src/ Or SourceForge: http://sourceforge.net/project/showfiles.php?group_id=139143&package_id=224… This is the first release coming out o SVN. Old CVS repository was migrated to SVN, and with these, a few changes apply to get sources from repository. Please read instructions from download page: http://www.openser.org/mos/view/Download/ Migration instructions (although not complete, I'm sure) are available at: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x The documentation for modules is posted at: http://openser.org/docs/modules/1.2.x/ We would like to thank to all developers and contributors for their work with coding and testing for this new release. Cheers, Daniel

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[Serusers] Header Selects
by Andrey Kuprianov 02 Apr '07

02 Apr '07

Hi, @msg.event or @msg.supported, or even @msg.["header-name"] is not implemented yet? im using latest ser from CVS.. btw, is it ok to use latest SER 2.0 for presence instead of the suggested snapshot v.4? I've noticed xcap-root and auth_xcap_root params were removed from rls and pa modules, so i've set xcap_root param in xcap module instead.

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[Users] presence logging....
by Helge Waastad 02 Apr '07

02 Apr '07

Hi, I'm just wondering if it's possible to turn off logging of some presence status messages? I have debug=0 But still I see a lot of: pr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: generate_ETag: etag= a.1175493921.28662.1 / 20 Apr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: get_subs_dialog:The query for subscribtion for [user]= xxx,[domain]= yyy for [event]= presence returned no result Apr 2 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE:query_db_notify: Could not get subs_dialog from database /32] 08:06:11 proxy-02 /usr/sbin/openser[28662]: PRESENCE: handle_publish: sip-etag: [SIP-ETag: a.1175493921.28662.1 Is the only way do only have L_ERR filtered in syslog daemon? br hw

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[Serusers] About avp_db_store()
by chungyu 02 Apr '07

02 Apr '07

Hi all, I am trying to use avp_db_store() stores my table "contact" with caller uri ex: table name: contact ----------------------------- | username | domain | ----------------------------- | | | ----------------------------- | | | ----------------------------- If (method=="INVITE"){ avp_write("$ruri/username", "s:name"); avp_db_store("$ruri", "s:name/contact"); }; if the incoming call then save username and domain of caller in contact caller: bob(a)iptel.org => ----------------------------- | username | domain | ----------------------------- | bob | iptel.org | ----------------------------- | | | ----------------------------- but not success, system reply ERROR:avpops:db_store: insert faild Please help me in the problem Thanks a lot in advance chungyu

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[Users] Call dropped after 6 ACKs. Branch=0 problem?
by kjcsb 02 Apr '07

02 Apr '07

I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, the call is established but often (but not always) eventually dropped. In the following dialogue, it can be seen that when the ACK is passed from the gateway on to Asterisk, OpenSER appends a Via with branch=0. After a number of these ACKs are passed on, the call is eventually dropped. It has been suggested by someone on the Asterisk list that "While "branch=0" is valid in RFC 2543, I don't think an INVITE can start-off as RFC 3261 and then the ACK can switch over to RFC 2543 in the middle of the transaction." i.e. the implication is that it's an OpenSER problem. Does that seem right? If so how can it be corrected? If not, could anyone advise what the problem might be or suggest any resolutions? U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 INVITE sip:6499777777@domain.co.nz SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 147.202.nnn.nnn:5060 "Noisy feedback tells: pid=29716 req_src_ip=202.180.nnn.nnn req_src_port=5060 in_uri=sip:6499777777@domain.co.nz out_uri=sip:6499777777@203.89.nnn.nnn via_cnt==1" U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 INVITE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 INVITE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 02 Apr 2007 02:53:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 8802 8802 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 35268 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16301 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16301 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK013cd54d;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK013cd54d;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK54b95a19;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK54b95a19;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK46a352b9;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK46a352b9;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK2da46c51;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK2da46c51;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK23a5accc;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK23a5accc;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK153c7273;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK153c7273;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK01e676a5;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK01e676a5;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd5ab.736d6085.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0b505d2b;rport=5060 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:6499777777@203.89.nnn.nnn> Content-Type: application/sdp Content-Length: 244 v=0 o=root 16300 16300 IN IP4 203.89.nnn.nnn s=session c=IN IP4 203.89.nnn.nnn t=0 0 m=audio 19952 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=silenceSupp:off - - - - U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0a0ddae8;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 ACK sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0a0ddae8;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 202.180.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 BYE sip:6499777777@203.89.nnn.nnn SIP/2.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 U 147.202.nnn.nnn:5060 -> 203.89.nnn.nnn:5060 BYE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as3f0f7d93;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKe5ab.311dd295.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 U 203.89.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKe5ab.311dd295.0;received=147.202.nnn.nnn Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 U 147.202.nnn.nnn:5060 -> 202.180.nnn.nnn:5060 SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK12646817;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as3f0f7d93 To: <sip:6499777777@domain.co.nz>;tag=as060f43a9 Call-ID: 2f573abd3cd80bd369495e322ec5ab4a(a)202.180.nnn.nnn CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Thanks Cameron

1 0

[Serusers] Unreacheable destination problems
by Amanda Mattiuz 01 Apr '07

01 Apr '07

Hello! I'm using SER + rtpproxy, but altough the NAT problem seems to be solved, I get no audio ... analysing the traffic with ethereal, I realized that the traffic going from the proxy to the UAs never arrives (ICMP unreacheable destination)... I assume it is because the users have their ports blocked, is that correct? I changed the rtpproxy port range (PORT_MIN and PORT_MAX ) to lower ports, but still I got the same messages... Does anybody knows anything that can be done by me in terms of configurations of the rtpproxy or ser.cfg or whatever, given the fact that I don't have power to open the ports??..... Thanks, Amanda -- ------------------------------------------------------------------- Amanda Remes Mattiuz amattiuz(a)gmail.com

1 0

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sr-users April 2007