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37519 discussions

[Users] Problem Radius Authentication - NO username in DIGEST-URI
by bogos dan-cristian 01 Dec '06

01 Dec '06

Hello Folks, I got some strange issue with RadAuth module. I have a running server which works just great with RadAuth. The problem comes when in the FROM and TO headers are no descriptions. EG: Following request will be properly processed: # INVITE sip:0012345@babble.net SIP/2.0. To: "0012345"<sip:0012345@babble.net>. From: "Some User"<sip:user@babble.net>;tag=5a7e9242. BUT the following one will not: INVITE sip:00123456@babble.net SIP/2.0. To: <sip:0012345@babble.net>. From: <sip:user@babble.net:10101>;tag=1-2963920204. Here is what I will get WRONG in RADIUS Server: Packet-Type = Access-Request Thu Nov 30 22:52:31 2006 User-Name = "user(a)babble.net" Digest-Attributes = "\n\020user(a)babble.net" Digest-Attributes = "\001\014babble.net" Digest-Attributes = "\002*456f61dc0d715ea80fd0ed0431df75b73a4c4092" Digest-Attributes = "\004\020sip:babble.net" Digest-Attributes = "\003\010INVITE" Digest-Response = "c96792da6dfb093215150ccd250fac34" Service-Type = Sip-Session Sip-Uri-User = "user" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Client-IP-Address = 87.102.50.17 The problem is that in the Digest-URI part I will not get the sip:0012345@babble.net, but the sip:babble.net, therefore completly erroneous info. I have checked with xlog the request on openser server just before passing the info to radius_proxy_authorize function, and all the FROM-URI, TO-URI, R-URI avps were just fine. Can somebody from the DEVELOPERS help me? I cannot see other solution, since there is nothing to be tuned in the settings of radauth module. Thank you in advance for your time! Cheers, Dan ____________________________________________________________________________________ Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com

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[Serusers] Download for later version of SER - "cvs head"?
by Paul Antinori (pantinor) 01 Dec '06

01 Dec '06

I have been using 0.9.6 version of SER that has DNS SRV load balancing, but now want to try the build that has DNS SRV failover/rollover capabilities built in using the SRV priority. I think this is called the "CVS HEAD" build. I cannot see this on the FTP download directories. Is this available? Thank you, Paul

3 2

[Users] Questions about regexp and AVPs
by Mahesh Paolini-Subramanya 30 Nov '06

30 Nov '06

I think I'm missing something pretty obvious here, but is there a reason that avp_check("$avp(tmpVar1)", "re/$avp(myNetwork0)/i") barfs on 1.1.x? I get the error 0(0) ERROR:avpops:fixup_check_avp: regexp operation requires string value 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 3491 In the above, $avp(tmpVar1) == "1.2.3.4" $avp(myNetwork0) == "1.2.3" Now, I'd hoped that I could actually run the following avp_check("$avp(tmpVar1)", "re/^$avp(myNetwork0)/i") but that doesn't work Then, setting $avp(myNetwork0) == "^1.2.3" and running avp_check("$avp(tmpVar1)", "re/$avp(myNetwork0)/i") didn't work either Sooo, whats the solution? avp_check can't have AVPs on both sides of the check? cheers -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

2 2

RE: [Users] regular expression
by Edson 30 Nov '06

30 Nov '06

First thinks first: please don't forget to CC the list.. Any information can be for others help. ;) Well, I would suggest You to make some tests on a command line simulation to be sure what You want to accoplish is even possible. Something like this: # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]@" # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]\@" # echo "sip:112345678#@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]" sip:112345678#@10.10.10.10 # echo "sip:112345678@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]\@" sip:112345678@10.10.10.10 # echo "sip:112345678@10.10.10.10" | grep -E "^sip:1[0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]@" sip:112345678@10.10.10.10 # As OpenSER, and SER as origin, use a regular expression as GREP, You could see that it work like You asked for. I also run a test in a SER leaving here. I put this code: } else if (uri=~"^sip:1[0-9]{8}@") { # Test for "1xxxxxxxx" mask numbers strip(9); prefix("600"); route(6); break; } And when I call 112345678 from my Softphone, ngrep shows me this forwarded INVITE: INVITE sip:600@pstn.gw:5060 SIP/2.0..Record-Route: <sip:sip.proxy;ftag=1468362515290;lr=on>..Via: SIP/2.0/UDP sip.proxy;branch=z9hG4bK90b1.5f c55385.0..Via: SIP/2.0/UDP client.ip;received=sip.client;rport=1024;branch=z9hG4bKac1bf8060000010f456ef 12a000028cf00000158..From: "Edson - SJPhone"<sip:8201@sip.proxy>;tag=1468362515290..To: <sip:112345678@sip.proxy>..Contact: <sip:8201@sip.client>..Call-ID: 647DD163-9B1E-45DB-93A2-99 CAC53532B8@client.ip..CSeq: 2 INVITE..Max-Forwards: 16..User-Agent: SJphone/1.61 (SJ Labs)..Content-Length: 217..Content-Type: application/sdp....v=0 ..o=- 3373887402 3373887402 IN IP4 sip.client..s=SJphone..c=IN IP4 sip.client..t=0 0..a=setup:active..m=audio 49186 RTP/AVP 4 101..a=rtpmap:4 G723/8000 ..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11,16.. Which in case is what we are looking for. Identified and translate any 9 digit number, started with '1' to '600'. Hope that this help You understand all the mechanism. Edson. _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: quinta-feira, 30 de novembro de 2006 01:24 To: Edson Subject: Re: [Users] regular expression Hello Edson, On 11/29/06, Edson <4lists(a)gmail.com> wrote: Opppsss.... I think that You make a little typo.... Wouldn't the correct be "^sip:1[0-9]{8}@" to match all numbers with 9 digits initiated by an '1' and followed by any combination of digits (from '0' to '9')? this Regular epression is sip:1[0-9]{8} correct but...... if the number like this ... 112345678# The Red color one must be the digit , not any any special charecter....... According u reg expression .. So,,, i'm sure ..................... like this it must .......... sip:1{8}[0-9}............ Sorry , if i did any thing mistake......excuse me......... Edson _____ From: users-bounces(a)openser.org [mailto: <mailto:users-bounces@openser.org> users-bounces(a)openser.org] On Behalf Of raviprakash sunkara Sent: quarta-feira, 29 de novembro de 2006 11:35 To: Jayesh Nambiar Cc: Users(a)openser.org Subject: Re: [Users] regular expression On 11/29/06, Jayesh Nambiar <voip_freak(a)yahoo.co.in <mailto:voip_freak@yahoo.co.in> > wrote: Hi all, Can someone please help me in formulating a regular expression. I have an expression like this: if(uri=~"^sip:1[0-9]*@) This matches numbers like 15552221212, but also matches number like 15552221212#. when i make it as: if(uri=~"^sip:1[0-9]@*) Use this if (uri=~"^sip:1{8}[0-9]@" ) Again this matches any special characters after the number. I just wanted a regular expression that will match only numbers and no alphabets or special characters. Can someone please help me to formulate a regular expression for this. Thanks in advance. w/regards, Jayesh _____ Find out what India is talking about on - Yahoo! Answers India <http://us.rd.yahoo.com/mail/in/yanswers/*http:/in.answers.yahoo.com/> Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW <http://us.rd.yahoo.com/mail/in/messengertagline/*http:/in.messenger.yahoo.c om> _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com <http://www.hyperion-tech.com> -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Serusers] error while b2b installation
by rupesh boyana 30 Nov '06

30 Nov '06

hi, after downloading latest b2b from vovida (sip-1.5.0.tar.gz), when i tried to install it, got the following error : In file included from Timer.cxx:58: Timer.h: In member function `bool Timer<T>::sleepFor(timeval*)': Timer.h:230: error: `cout' was not declared in this scope Timer.h:230: error: `endl' was not declared in this scope Timer.h: In member function `void Timer<T>::insert(T, int)': Timer.h:267: error: `cout' was not declared in this scope Timer.h:267: error: `endl' was not declared in this scope make[1]: *** [obj.debug.Linux.i686/Timer.o] Error 1 make[1]: Leaving directory `/root/sip-1.5.0/util' make: *** [util] Error 2 Initially i gave ./configure inside the sip-1.5.0 folder. Then when i tried with make b2bua , got the following error and installation stopped. i have ser with mysql,nathelper,rtp proxy. thank you, Rupesh.

2 1

[Users] auth_db subscriber caching
by Klaus Darilion 30 Nov '06

30 Nov '06

Hi! A stupid question: Is the subscriber table also cached inside openser (like location table)? regards klaus -- Klaus Darilion nic.at

2 5

[Serusers] SER/Asterisk inbound problem [corrected]
by Sameer Ahmed 30 Nov '06

30 Nov '06

The Setup: I have two asterisk servers, AsterA and AsterB, and one ser server acting as a proxy. Both AsterA and AsterB are connected to Ser as sip clients. On the asterisk servers I have defined a SIP trunk. The idea is that users/extensions on AsterA should be able to reach users/extensions on AsterB via Ser. The Problem: My extension numbers on AsterA range from 201 to 220. On AsterB I also have the same range of extensions, 201 to 220. When user 205 on AsterA places a call to user 201 on AsterB, he gets a network congestion message. Looking at the sip packets in Asterisk CLI, I get a "Forbidden" message from AsterB. Here's the interesting bit: When I create a user/extension on AsterA that does not exist on AsterB, like 401, I do not get this problem and the call goes through just fine. Apparently, the problem has to do with the fact that I am using the same list of extensions on both asterisk servers. Detailed Configuration: All three servers are on the same network. IP Address of AsterA: 192.168.0.9 IP Address of AsterB: 192.168.0.14 IP Address of SER: 192.168.0.15 To dial out from either asterisk server using the SIP trunk: 91 must be entered before the number SER must know which server the call is meant for, so the following codes are used by SER to distinguish between calls for AsterA and AsterB: AsterA: 52 AsterB: 51 Example 1: If 205 on AsterA wants to call 201 on AsterB, he must dial 9151201 Example 2: If 201 on AsterB wants to call 205 on AsterA, he must dial 9152205 I have insecure=invite in both sip.conf files, for AsterA and AsterB Below are the sip packets I picked up at AsterA (192.168.0.9) when 9152201 was dialed by extension 205 at AsterB (192.168.0.14) <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15 ;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: < sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9000 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 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 - - - - --- (15 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1371cb5b" Content-Length: 0 --- Scheduling destruction of call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14' in 15000 ms Found user '205' <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: < sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 102 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15 ;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: < sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Proxy-Authorization: Digest username="bf51", realm="asterisk", algorithm=MD5, uri="sip:52201@192.168.0.15", nonce="1371cb5b", response="8ecd1a24c31a02fc938308a0f5c48924", opaque="" Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9001 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 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 - - - - --- (16 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Found user '205' Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15 ;branch=z9hG4bKcb71.18308a81.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 103 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14'

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[Serusers] SER/Asterisk inbound problem
by Sameer Ahmed 30 Nov '06

30 Nov '06

The Setup: I have two asterisk servers, AsterA and AsterB, and one ser server acting as a proxy. Both AsterA and AsterB are connected to Ser as sip clients. On the asterisk servers I have defined a SIP trunk. The idea is that users/extensions on AsterA should be able to reach users/extensions on AsterB via Ser. The Problem: My extension numbers on AsterA range from 201 to 220. On AsterB I also have the same range of extensions, 201 to 220. When user 205 on AsterA places a call to user 201 on AsterB, he gets a network congestion message. Looking at the sip packets in Asterisk CLI, I get a "Forbidden" message from AsterB. Here's the interesting bit: When I create a user/extension on AsterA that does not exist on AsterB, like 401, I do not get this problem and the call goes through just fine. Apparently, the problem has to do with the fact that I am using the same list of extensions on both asterisk servers. Detailed Configuration: All three servers are on the same network. IP Address of AsterA: 192.168.0.9 IP Address of AsterB: 192.168.0.14 IP Address of SER: 192.168.0.15 To dial out from either asterisk server using the SIP trunk: 91 must be entered before the number SER must know which server the call is meant for, so the following codes are used by SER to distinguish between calls for AsterA and AsterB: AsterA: 51 AsterB: 52 Example 1: If 205 on AsterA wants to call 201 on AsterB, he must dial 9152201 Example 2: If 201 on AsterB wants to call 205 on AsterA, he must dial 9151205 I have insecrure=invite in both sip.conf files, for AsterA and AsterB Below are the sip packets I picked up at AsterA (192.168.0.9) when 9152201 was dialed by extension 205 at AsterB (192.168.0.14) <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: <sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9000 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 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 - - - - --- (15 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1371cb5b" Content-Length: 0 --- Scheduling destruction of call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14' in 15000 ms Found user '205' <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 102 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: <sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Proxy-Authorization: Digest username="bf51", realm="asterisk", algorithm=MD5, uri="sip:52201@192.168.0.15", nonce="1371cb5b", response="8ecd1a24c31a02fc938308a0f5c48924", opaque="" Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9001 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 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 - - - - --- (16 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Found user '205' Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 103 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14'

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RE: [Users] Multiple register message
by Thomas Deillon 30 Nov '06

30 Nov '06

Thanks a lot. It's work. Thomas -----Message d'origine----- De : Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Envoyé : mercredi, 29. novembre 2006 19:24 À : Thomas Deillon Cc : Klaus Darilion; users(a)openser.org Objet : Re: [Users] Multiple register message Hi Thomas, your script will not correctly work as it is a combination of stateless and stateful processing. t_replicate, t_on_failure() are stateful functions, while forward() does stateless fwd - try t_relay() from TM module. regards, bogdan Thomas Deillon wrote: >Subject: Trying to find a solution to duplicate an incoming REGISTER to >two asterisks boxes. > >1. maybe Asterisk can use the location table of openser too? >2. you can save the contact with save_no_reply and then t_relay to the >asterisk server, or > >1&2 > In fact, SER didn't register phones. It just make forwards > >3. you can use t_replicate ... > > >>I tried but I didn't find the way to do this. >> >> > >I have the configuration below : > > >route{ > > if(method=="REGISTER"){ > t_replicate("192.168.1.119") > forward("192.168.1.120") > } > > ds_select_dst("0", "4"); > forward(); > t_on_failure("1"); >} > >failure_route[1] { > log("next_dst\n"); > ds_next_dst(); >} > > >Thanks a lot for your help. >Thomas > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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[Serusers] how to get source uri?
by jorain 30 Nov '06

30 Nov '06

Hi all, How to get the source uri from exec function? like SIP_USER, SIP_RURI and so on. SIP_HF_FROM give full path, including source username, uri and tag. I just want to get the source uri. Please guide. Thanks for any help. Regards, jorain

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