- sr-users - kamailio.org

[Users] Method of tracking dead accounts?
by Nick De Cristofaro 15 Feb '07

15 Feb '07

We are looking to find a way to audit the accounts on our SER that haven't been used in a long time so we can decide if they are dead and we can remove them, hence re-use their PSTN numbers. I'm not sure if there's any module built in that could do this? Any ideas? There's 4000 accounts approximetely. Thanks Nick

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[Serusers] user aliases in ottendorf??
by tzieleniewski 15 Feb '07

15 Feb '07

Hi !! Is there still some mechanism for user aliases in ottendorf?? I ask because when I add a user through serweb it adds two records to the uri table one for a canonical name and the other for numeric address. What has to be done (what function called in ser.cfg) so that whenever user will be called on that numerical name there will be a possibility to forward the call to him, like it was previously done in ser 0.9 with lookup("aliases")? Bests -Tomasz

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Re: [Users] codec translation
by vallimamod abdullah 15 Feb '07

15 Feb '07

Hello, On Thu, 15 Feb 2007 11:19:37 +0100, Atle Samuelsen <clona(a)cyberhouse.no> wrote: > Hi Guys, > > Just qurious.. How intrested are you guys in a very simple transcoder, > wich is based on rtpproxy ? > I would be really interested in a g729 to g711 transcoder as I have a pstn gateway that doesn't support g729. I am currently using mediaproxy but I would consider to either move to rtpproxy with transcoder functionality or contribute to implement this feature in mediaproxy. -- Regards, -vma .

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[Users] not install mysql.so
by Javier Allende Astigarraga 15 Feb '07

15 Feb '07

i try to install openser from tar.gz and i do it make all include_modules="mysql" make prefix=/usr/local install include_modules="mysql" But dont appears mysql.so what the matter?? thanks por all -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

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[Users] codec translation
by tusker keg 15 Feb '07

15 Feb '07

Howdy I have a situation I hope you guys will help me out with I am receiving call from a VOIP peer (SIP Call) and the peer can only send them as G711. I need to redirect to call to another voip peer over the wan and due to bandwidth considerations I need to translate the codec to g729. Any ideas on how to do this, Sample config file will be help Regards ./Tusker

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[Serusers] lookup_contacts() error after sd_lookup() from the speeddial module- latest cvs ottendorf
by tzieleniewski 15 Feb '07

15 Feb '07

Hi! I have encountered the following situation after I make the translation from the speeddial number into the proper uri the user cannot be found by the lookup_contacts() function. Calling a user directly works without any problem: My logic is the following: dump_attrs(); if (uri =~ "sip:[0-9]{2}@.*") { sd_lookup("speed_dial"); } dump_attrs(); if (lookup_user("$t.uid", "@ruri")) { dump_attrs(); load_attrs("$tu", "@ruri"); load_attrs("$tr", "@ruri"); dump_attrs(); if (lookup_contacts("location")) { append_hf("P-hint: usrloc applied\r\n"); route(FORWARD); drop; } else { route(CALL_FORWARD_VOICEMAIL); drop; } } Here is the debug output: Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: <INVITE> <sip:01@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (uri =~ sip:[0-9]{2}@.*) <INVITE> <sip:01@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: sd_lookup(speed_dial) Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (lookup_user($t.uid, @ruri)) <INVITE> <sip:hellboy@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: load_attrs($tu,@ruri) <INVITE> <sip:hellboy@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: load_attrs($tr,@ruri) <INVITE> <sip:hellboy@voip.touk.pl> Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:540: class=GLOBAL Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:560: track=TO class=DOMAIN Feb 15 14:15:23 rd ser[14469]: AVP["did"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["digest_realm"]="voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:570: track=FROM class=USER Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["email"]="tzl(a)touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["lang"]="en" Feb 15 14:15:23 rd ser[14469]: AVP["tyk"]="tyk" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:580: track=TO class=USER Feb 15 14:15:23 rd ser[14469]: INFO: No AVP present Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:590: track=FROM class=URI Feb 15 14:15:23 rd ser[14469]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: INFO: avp.c:600: track=TO class=URI Feb 15 14:15:23 rd ser[14469]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: AVP["forward_blindd"]="sip:evn@voip.touk.pl" Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: (lookup_contacts(location)) Feb 15 14:15:23 rd ser[14469]: route[INBOUND]: route(CALL_FORWARD_VOICEMAIL) <INVITE> <sip:hellboy@voip.touk.pl> Bests -tomasz

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Re: [Serusers] Inoltra: Problems with Rtpproxy traffic balancing in SER 0.10.x
by Ladislav Andel 15 Feb '07

15 Feb '07

what is written in your modparam rtpproxy line? how do you start your rtpproxy? (what parameters do you use) There might be some changes in the code as well since the document was issued few months ago and the development of SER is going on. I'm trying to reproduce it with the latest CVS version.. Lada Angelo Taddeo wrote: > I've done correctly anything you wrote in your mail. > > I've found my mistake with $node: in database i put it with flag 32 > instead of 33. > > Anyway one problem remains: it isn't enough inserting rtpproxy IP > address in modparam (in ser.cfg). > If I dont't give this IP to function/ force_rtp_proxy/ in ser.cfg it > doesn't work, that is it doesn't write correctly rtpproxy IP in > connection information (SDP) during SDP rewriting . > I've looked at code. SER expects rtpproxy IP address in reply from > rtpproxy. But rtpproxy in its message to SER doesn't write its IP!!!!! > > So in my ser.cfg I have to define a variable conteining rtpproxy IP > address and I have to give it to function force_rtp_proxy. Only in > this way it works. > > I wait for your replay. > > I'm using SER 10.0.x CVS version! > > Thanks in advance > > Angelo > > > > > */Ladislav Andel <ladaan(a)iptel.org>/* ha scritto: > > It is saved in database table "user_attrs". Look at the page 24 in > optimized_nat_traversal.pdf > document how it > should look like. > UID in that table means username. > The value of variable "node" means the index of RTPproxy starting > from 0. > For example: > modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock > udp:5.6.7.8:22222") > > unix:/var/run/rtpproxy.sock is O > udp:5.6.7.8:22222 is 1 > > Within your ser.cfg you load these AVP variables through > load_attrs(); > and use them in nat mangle route. > You can check the value of "node" by dump_attrs(); before calling > force_rtp_proxy and see if there is correct value. > > Lada > > > > Angelo Taddeo wrote: > > > > > > > > Hi again > > I'm using ser 0.10.x ( CVS version). > > > > About rtpproxy traffic balancing: > > > > Where I have to put <$node> value???????? > > > > Thanks > > > > > ------------------------------------------------------------------------ > > L'email della prossima generazione? Puoi averla con la nuova Yahoo! > > Mail > > > > ------------------------------------------------------------------------ > L'email della prossima generazione? Puoi averla con la nuova Yahoo! > Mail > <%20http://us.rd.yahoo.com/mail/it/taglines/hotmail/nowyoucan/nextgen/*%20http://it.docs.yahoo.com/nowyoucan.html>

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Re: [Users] radius authentication
by Stefano Capitanio 15 Feb '07

15 Feb '07

Hi, i've configured also "servers" files. i've also set: modparam("auth_radius", "service_type", 6) but i cannot set: modparam("auth_radius","Sip-Session",15), it says "parameter <Sip-Session> not found in module <auth_radius>" my dictionary file is: #### Attributes ### ATTRIBUTE User-Name 1 string # RFC2865, acc, auth_radius, avp_radius, group_radius, uri_radius ATTRIBUTE Service-Type 6 integer # RFC2865, acc, auth_radius, avp_radius, group_radius, uri_radius #ATTRIBUTE Called-Station-Id 30 string # RFC2865, acc #ATTRIBUTE Calling-Station-Id 31 string # RFC2865, acc #ATTRIBUTE Acct-Status-Type 40 integer # RFC2865, acc #ATTRIBUTE Acct-Session-Id 44 string # RFC2865, acc ATTRIBUTE Sip-Method 101 integer # Schulzrinne, acc ATTRIBUTE Sip-Response-Code 102 integer # Schulzrinne, acc ATTRIBUTE Sip-Cseq 103 string # Schulzrinne, acc ATTRIBUTE Sip-To-Tag 104 string # Schulzrinne, acc ATTRIBUTE Sip-From-Tag 105 string # Schulzrinne, acc ATTRIBUTE Sip-Translated-Request-URI 107 string # Proprietary, acc ATTRIBUTE Digest-Response 206 string # Sterman, auth_radius ATTRIBUTE Sip-Uri-User 208 string # Proprietary, auth_radius ATTRIBUTE Sip-Group 211 string # Proprietary, group_radius ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius ATTRIBUTE SIP-AVP 225 string # Proprietary, avp_radius ATTRIBUTE Digest-Realm 1063 string # Sterman, auth_radius ATTRIBUTE Digest-Nonce 1064 string # Sterman, auth_radius ATTRIBUTE Digest-Method 1065 string # Sterman, auth_radius ATTRIBUTE Digest-URI 1066 string # Sterman, auth_radius ATTRIBUTE Digest-QOP 1067 string # Sterman, auth_radius ATTRIBUTE Digest-Algorithm 1068 string # Sterman, auth_radius ATTRIBUTE Digest-Body-Digest 1069 string # Sterman, auth_radius ATTRIBUTE Digest-CNonce 1070 string # Sterman, auth_radius ATTRIBUTE Digest-Nonce-Count 1071 string # Sterman, auth_radius ATTRIBUTE Digest-User-Name 1072 string # Sterman, auth_radius ### CISCO Vendor Specific Attributes ### #VENDOR Cisco 9 #ATTRIBUTE Cisco-AVPair 1 string Cisco # VSA, auth_radius ### Acct-Status-Type Values ### #VALUE Acct-Status-Type Start 1 # RFC2866, acc #VALUE Acct-Status-Type Stop 2 # RFC2866, acc VALUE Acct-Status-Type Failed 15 # RFC2866, acc ### Service-Type Values ### VALUE Service-Type Call-Check 10 # RFC2865, uri_radius VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 # Schulzrinne, acc, auth_radius VALUE Service-Type SIP-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type SIP-Callee-AVPs 31 # Proprietary, avp_radius any idea? raviprakash sunkara ha scritto: > Can U Check the servers in radiusclient dir. > That host and key should match the Radiu server NASlist file or nas > tables and client.conf file > > modparam("auth_radius","Sip-Session",15); > > On 2/15/07, *Stefano Capitanio* <s.capitanio(a)caspur.it > <mailto:s.capitanio@caspur.it>> wrote: > > Ciao! > > I'm trying to authenticate using Radius. > I've installed radiusclient-ng 0.5.2 > i've configured radiusclient.conf with the IP address of the > radius server. > I've configured SER with: > ... > modparam("auth_radius", "radius_config", > "/etc/radiusclient-ng/radiusclient.conf") > ... > if (method=="REGISTER") { > if (!radius_www_authorize(" example.it <http://example.it>")) { > www_challenge("example.it > <http://example.it>", "0"); > exit; > } > save("location"); > } > > but it doesn't work: there seem to be a problem in the client because > there is no IP Packet that goes out directed to the radius server... > in the log of SER i have no errors... > > thank you for any help, > Stefano > > _______________________________________________ > Users mailing list > Users(a)openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > www.hyperion-tech.com <http://www.hyperion-tech.com> > Client and Parent company :- www.august-networks.com > <http://www.august-networks.com>

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[Serusers] mediaproxy exceptions on session delete
by Klaus Darilion 15 Feb '07

15 Feb '07

Hi! I have a problem with mediaproxy 1.8. Whenever a session is deleted there is an exception: Feb 14 14:18:27 foobar mediaproxy[27053]: delete OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU. info= Feb 14 14:18:27 foobar mediaproxy[27053]: session OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU.: 10558/10557/21113 packets, 778927/774750/1553531 bytes (caller/called/relayed) Feb 14 14:18:27 foobar mediaproxy[27053]: session OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU.: ended. Feb 14 14:18:27 foobar mediaproxy[27053]: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected 5.6.7.8:38879 at 0xb7d255ec> (exceptions.NameError:global name 'StopRecordSerializer' is not defined [/usr/lib/python2.3/asyncore.py|readwrite|86] [/usr/lib/python2.3/asyncore.py|handle_read_event|390] [/opt/mediaproxy/modules/rtphandler.py|handle_read|441] [/opt/mediaproxy/modules/rtphandler.py|process|568]) Feb 14 14:19:53 foobar mediaproxy[27053]: status Feb 14 14:19:53 foobar mediaproxy[27053]: execution time: 0.11 ms I could not find the problem - accounting.py with StopRecordSerializer is there, and it is imported in rtphandler. Do others also have problems and maybe a workaround? regards klaus -- Klaus Darilion nic.at

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[Serusers] TCP main process and TCP receiver not starting
by Andrew T Gin 15 Feb '07

15 Feb '07

Hi I'm using the Hello World.cgf as my configuration file. My problem is that SER does not appear to be listening for TCP connections; it returns an RST when it receives a SYN. at http://mit.edu/sip/sip.edu/ser.shtml it says a correctly running SER listening on 10.1.2.3 and the loop back configured to use 4 processes looks like this: # serctl ps 0 25254 attendant 1 25258 receiver child=0 sock=0 @ 10.1.2.3:5060 2 25259 receiver child=1 sock=0 @ 10.1.2.3:5060 3 25261 receiver child=2 sock=0 @ 10.1.2.3:5060 4 25263 receiver child=3 sock=0 @ 10.1.2.3:5060 5 25265 receiver child=0 sock=1 @ 127.0.0.1:5060 6 25267 receiver child=1 sock=1 @ 127.0.0.1:5060 7 25274 receiver child=2 sock=1 @ 127.0.0.1:5060 8 25280 receiver child=3 sock=1 @ 127.0.0.1:5060 9 25282 fifo server 10 25312 timer 11 25314 tcp receiver 12 25316 tcp receiver 13 25318 tcp receiver 14 25320 tcp receiver 15 25323 tcp main process However when I run serctl ps I get the following: $ serctl ps 0 31360 stand-alone receiver @ 132.181.14.210:5060 1 31361 timer 2 31362 fifo server 3 0 4 0 5 0 6 0 7 0 This SER is listing on 132.181.14.210, but there is only a stand-alone receiver, not receiving childs. I have specified in the cfg file children=4, but it doesnt appear to do so. What am I missing, and why don't the tcp receivers, receiver childs and tcp main process start? Thanks Andrew

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[Serusers] problem with status in serial forking (maybe related to bug 48)
by Zappasodi Daniele 15 Feb '07

15 Feb '07

Hello, I have a problem with serial forking and it seems the same behaviour of the bug 48: my ser.cfg script stops the first call due to time-out (fr_inv_timer), and the status in the failure_route is 408, and then sends a call to a second destination. If the caller hang up the call, the status in failure-route should be 487, but I have 408 again. The bug is declared as closed since the 0.9.5 (http://tracker.iptel.org/browse/SER-48), where can I take the patch? I have downloaded the version 0.9.6 from http://ftp.iptel.org/pub/ser/0.9.6/src/, but it doesn't seem to contain the patch for the bug. thanks ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

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[Serusers] Inoltra: Problems with Rtpproxy traffic balancing in SER 0.10.x
by Angelo Taddeo 15 Feb '07

15 Feb '07

Hi again I'm using ser 0.10.x ( CVS version). About rtpproxy traffic balancing: Where I have to put <$node> value???????? Thanks --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail

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[Serusers] Routing based on FROM
by Edson 15 Feb '07

15 Feb '07

Hi, Guys... I'm challenged to implement a routing that consider the FROM header. That is, if call goes to X and comes from user U, then route to GW G1, else route to GW2. In version 0.9.6, how to find the "from user" value? Edson

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[Serusers] Re: Serusers Digest, Vol 43, Issue 12
by wangdq＠gongye.com.cn 15 Feb '07

15 Feb '07

test On Sun, 2006-11-12 at 12:00 +0000, serusers-request(a)lists.iptel.org wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. SER not sending 486 Busy Here (Ravi *****) > 2. Re: SER pre-release - PRESENCE ISSUES (SER LIST) > 3. Suggestion (Eddie Johnson Jr) > 4. SER pre-release - Presence and NOTIFY (SER LIST) > 5. SER pre-release ser_ctl (SER LIST) > 6. Re: SER pre-release ser_ctl (Alexandr Dubovikov) > 7. Re: SER pre-release - PRESENCE ISSUES (Michal Matyska) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 11 Nov 2006 14:15:03 +0200 > From: "Ravi *****" <mravikreddy(a)gmail.com> > Subject: [Serusers] SER not sending 486 Busy Here > To: serusers(a)lists.iptel.org > Message-ID: > <cfbb389b0611110415l33701327vc3fd09fd9d571e10(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Users , > > I am Using SER-0.96 , When I make a call from > SER to Pstn it works fine I mean is all "INV"BYE"CAN". > > But when I recieved a call from Pstn I cannot cancel the call because > X-lite is sending to pstn gateway 480 temp. service not available instead > of 486 Busy Here. > > I am using default config. but I dont understand where > it went wrong . > > Any Idea or help greatly appreciated > > Thank You. > Regards, > Ravi. >

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[Serusers] Re: Serusers Digest, Vol 43, Issue 12
by wangdq＠gongye.com.cn 15 Feb '07

15 Feb '07

On Sun, 2006-11-12 at 12:00 +0000, serusers-request(a)lists.iptel.org wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. SER not sending 486 Busy Here (Ravi *****) > 2. Re: SER pre-release - PRESENCE ISSUES (SER LIST) > 3. Suggestion (Eddie Johnson Jr) > 4. SER pre-release - Presence and NOTIFY (SER LIST) > 5. SER pre-release ser_ctl (SER LIST) > 6. Re: SER pre-release ser_ctl (Alexandr Dubovikov) > 7. Re: SER pre-release - PRESENCE ISSUES (Michal Matyska) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 11 Nov 2006 14:15:03 +0200 > From: "Ravi *****" <mravikreddy(a)gmail.com> > Subject: [Serusers] SER not sending 486 Busy Here > To: serusers(a)lists.iptel.org > Message-ID: > <cfbb389b0611110415l33701327vc3fd09fd9d571e10(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Users , > > I am Using SER-0.96 , When I make a call from > SER to Pstn it works fine I mean is all "INV"BYE"CAN". > > But when I recieved a call from Pstn I cannot cancel the call because > X-lite is sending to pstn gateway 480 temp. service not available instead > of 486 Busy Here. > > I am using default config. but I dont understand where > it went wrong . > > Any Idea or help greatly appreciated > > Thank You. > Regards, > Ravi. >

1 0

[Serusers] Forcing Account
by Gustavo Passos Tourinho 14 Feb '07

14 Feb '07

Hello Guys Is there a way to force proxy accounting on radius without/before use t_relay()? My problem is: I have a SIP URI that is received by the proxy but should change that URI before forward it. For example: Proxy receive an INVITE: sip:123@teste.com Proxy should send leg to radius Proxy change uri to: sip:098@example.com Proxy shoud foward INVITE (t_relay) I saw acc_rad_request ("forced") on ser.cfg, but Im not sure about what it does. Thanks for any help. Gustavo

1 0

[Users] nathelper and flag 2
by Andreas Granig 14 Feb '07

14 Feb '07

Hi, Today I found a UAC which is *not* located behind NAT (public IP 1.2.3.4) and sends this Via-Header, which seems perfectly valid according to RFC3261: SIP/2.0/UDP VINNASUP06C:5060;maddr=1.2.3.4;branch=z9hG4bK-2198d2 I used to check for nated clients using nat_uac_test("3"), which detects NAT in this case, because the host-part doesn't match the received-address. So is the test-flag "2" useless, since the host-part can be "hostname / IPv4address / IPv6reference", or should this particular test be extended to also check for the maddr-parameter? In the meanwhile, I've changed my nat-test to "17" for only testing Contact and Via-Port instead of Contact and Via-Address, but it's still not optimal. Any opinions on this? Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation.

3 8

[Users] imc
by henri keski-sikkila 14 Feb '07

14 Feb '07

Hello, When I invite new member to instance message conference I get the message that doesn't contain header: Content-Type: text/plain This couses me little bit problems...Should this header include to the reply? MESSAGE sip:xx SIP/2.0. Via: SIP/2.0/UDP xx;branch=z9hG4bK74fb.067ba117.0. To: sip:xx. From:sip:x;tag=0c3c711902a8bbb20a1e767e10f39a78-96ab. CSeq: 10 MESSAGE. Call-ID: 7bdbafcb-32586@x. Content-Length: 38. User-Agent: OpenSer (1.2.0-dev6-notls (i386/linux)). . x@x joined room.. left Best Regards, Henri Keski-Sikkila

3 6

[Users] Makefile error on arm-linux OS
by Michel Bensoussan 14 Feb '07

14 Feb '07

Hello I'm trying to compile OpenSER for an arm platform (Linux). The compiler is arm-linux-gcc I tried the following command: *$ make CC=arm-linux-gcc* And there is the following errors: *cc1: error: invalid option `inline-all-stringops' cc1: error: bad value (athlon) for -mcpu= switch * The arm-linux-gcc -v gives: *Reading specs from /usr/local/lib/gcc-lib/arm-linux/3.3.3/specs Configured with: ../configure --target=arm-linux --disable-shared --disable-threads --with-gnu-as --with-gnu-ld --enable-multilib --enable-languages=c Thread model: single gcc version 3.3.3 * I also have to modified the libs. Is there a simple way to do this porting or do I have to check all compilation and link options and modified or re-write the Makefiles? Regards, Michel

1 0

[Serusers] [PRESENCE]-Subscription does not exist
by SER LIST 14 Feb '07

14 Feb '07

Hello- I am running CVS version of ser w/ presence. All seems to be working well, except that I see a lots of packets being returned by SER indicating "Subscription does not exist". SIP/2.0 481 Subscription does not exist. From: <sip:111@xx.com>;tag=3ffe8a226de90fb471c542cb1065d20a. To: <sip:111@xx.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-9321. Call-Id: ssc-3d959a71f819794a430784374ee6f664@4bfc7569f5fad1a1a28a85dcb86badbb. Cseq: 11 SUBSCRIBE. Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK01ee.cbc45cb3.0. Via: SIP/2.0/UDP 192.168.1.3:5065;received=67.X.X.X;branch=z9hG4bK-ecfc3a30c3920b9e;rport=5065. Server: Sip EXpress router (0.10.99-dev57-tm_fixes (i386/linux)). Content-Length: 0. These are in response to a SUBSCRIBE request with Event: presence.winfo - The packet above was sent by SER in response to: SUBSCRIBE sip:X.X.X.X:5060 SIP/2.0. Record-Route: <sip:X.X.X.X;ftag=3ffe8a226de90fb471c542cb1065d20a;lr>. From: <sip:111@xx.com>;tag=3ffe8a226de90fb471c542cb1065d20a. To: <sip:111@xx.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-9321. Call-Id: ssc-3d959a71f819794a430784374ee6f664@4bfc7569f5fad1a1a28a85dcb86badbb. Cseq: 11 SUBSCRIBE. Contact: sip:111@sip.tpad.com. Event: presence.winfo. Expires: 600. Date: Tue, 13 Feb 2007 21:18:18 GMT. Max-Forwards: 16. User-Agent: SomUserAgent Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, SUBSCRIBE, NOTIFY, MESSAGE. Supported: replaces. Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK01ee.cbc45cb3.0. Via: SIP/2.0/UDP 192.168.1.3:5065;received=67.X.X.X;branch=z9hG4bK-ecfc3a30c3920b9e;rport=5065. Content-Length: 0. P-hint: Loose Routed. Any suggestions? Thanks in advance? _________________________________________________________________ Buy what you want when you want it on Sympatico / MSN Shopping http://shopping.sympatico.msn.ca/content/shp/?ctId=2,ptnrid=176,ptnrdata=08…

2 1

[Users] help-me in 800
by Jobson Andrade 13 Feb '07

13 Feb '07

Hi All, I,m use openser 1.0.1 We imagined the following setting: 1- It Receive the connections of a pstn that sends all of the requisition for my one openser 2- it authenticate the coming calls of that IP 3- It Receive the PIN, authenticate and direct the call for the routes The question is like do that verification of the PIN´s and send the calls after that verification It plan is: It receives itself 0800 123 123 of an ONLY IP, through the src_ip==xxx.xxx.xxx.xxx sends for a route I obtained to remove with the strip I number him 0800 123 123 and with the prefix (""); send with the rewritehostport for a route I need that be attended, authenticated the PIN and receive of the side "A" I number of fate and sent for the route of smaller cost How to made this???? Please help-me!! Thanks in advanced!! Jobson Andrade Projetos & Desenvolvimento Obelisk - The Asterisk & VoIP Experts phone/fax: (11) 2164-4808 ext. 115 cell Phone: (11) 8175-9916 / 8271-0480 email: <mailto:jandrade@obelisknet.com.br> jandrade(a)obelisknet.com.br

1 0

[Users] script to create dbtext tables
by Cesc 13 Feb '07

13 Feb '07

Hi, I see there are scripts to automatically create the tables for mysql and postgres ... Does an equivalent tool exist for dbtext? Cesc

2 2

Re: Re: Re: Re: Re: Re: Re: [Serusers] call forwarding issues - load_attrs and lookup_contacts - ottendorf
by tzieleniewski 13 Feb '07

13 Feb '07

I did some tests. I sent them on the list because I wat not sure how to interpret their correctness beefore putting them on the tracker I did the following general procedure: dump_attrs() lookup_user(..) dump_attrs() load_attrs(..) dump_attrs() lookup_contacts(..) Case one: dump_attrs() lookup_user("$t.uid","@ruri") dump_attrs() load_attrs("$tu","$t.uid") dump_attrs() lookup_contacts("location") my observations: after lookup_user("$t.uid","@ruri") uid avp appered in the uri class and to track, lookup_contacts("location") found the user debug log: Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:540: class=GLOBAL Feb 13 13:48:25 rd ser[9875]: AVP["lang"]="en" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:570: track=FROM class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:580: track=TO class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:590: track=FROM class=URI Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:600: track=TO class=URI Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: route[INBOUND]: (lookup_user($t.uid, @ruri)) Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:540: class=GLOBAL Feb 13 13:48:25 rd ser[9875]: AVP["lang"]="en" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:570: track=FROM class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:580: track=TO class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:590: track=FROM class=URI Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:600: track=TO class=URI Feb 13 13:48:25 rd ser[9875]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: route[INBOUND]: load_attrs($tu,$t.uid) Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:540: class=GLOBAL Feb 13 13:48:25 rd ser[9875]: AVP["lang"]="en" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 13:48:25 rd ser[9875]: AVP["did"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: AVP["digest_realm"]="voip.touk.pl" Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:570: track=FROM class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:25 rd ser[9875]: INFO: avp.c:580: track=TO class=USER Feb 13 13:48:25 rd ser[9875]: INFO: No AVP present Feb 13 13:48:26 rd ser[9875]: INFO: avp.c:590: track=FROM class=URI Feb 13 13:48:26 rd ser[9875]: INFO: No AVP present Feb 13 13:48:26 rd ser[9875]: INFO: avp.c:600: track=TO class=URI Feb 13 13:48:26 rd ser[9875]: AVP["uid"]="hellboy(a)voip.touk.pl" Case two: dump_attrs() lookup_user("Request-uri") dump_attrs() load_attrs("$tu","$tu.uid") dump_attrs() lookup_contacts("location") my observations: after lookup_user("Request-uri") uid avp appered in the user class and to track, lookup_contacts("location") didn't find the user Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:540: class=GLOBAL Feb 13 14:22:57 rd ser[10047]: AVP["lang"]="en" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:580: track=TO class=USER Feb 13 14:22:57 rd ser[10047]: INFO: No AVP present Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:22:57 rd ser[10047]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:600: track=TO class=URI Feb 13 14:22:57 rd ser[10047]: INFO: No AVP present Feb 13 14:22:57 rd ser[10047]: route[INBOUND]: (lookup_user(Request-uri) Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:540: class=GLOBAL Feb 13 14:22:57 rd ser[10047]: AVP["lang"]="en" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:580: track=TO class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["ruri_canonical"]="1" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:22:57 rd ser[10047]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:600: track=TO class=URI Feb 13 14:22:57 rd ser[10047]: INFO: No AVP present Feb 13 14:22:57 rd ser[10047]: route[INBOUND]: load_attrs($tu,$t.uid) Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:540: class=GLOBAL Feb 13 14:22:57 rd ser[10047]: AVP["lang"]="en" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:22:57 rd ser[10047]: AVP["did"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:580: track=TO class=USER Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["ruri_canonical"]="1" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:22:57 rd ser[10047]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:22:57 rd ser[10047]: INFO: avp.c:600: track=TO class=URI Feb 13 14:22:57 rd ser[10047]: INFO: No AVP present Case three: dump_attrs() lookup_user("$tu.uid","@ruri") dump_attrs() load_attrs("$tu","$tu.uid") dump_attrs() lookup_contacts("location") my observations: after lookup_user("$tu.uid","@ruri") uid avp appered in the user class and to track, lookup_contacts("location") didn't find the user I also tried to invoke lookup_contacts("location","$tu.uid") but with the same result user wasn't found. Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:540: class=GLOBAL Feb 13 14:36:42 rd ser[10100]: AVP["lang"]="en" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:580: track=TO class=USER Feb 13 14:36:42 rd ser[10100]: INFO: No AVP present Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:36:42 rd ser[10100]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:600: track=TO class=URI Feb 13 14:36:42 rd ser[10100]: INFO: No AVP present Feb 13 14:36:42 rd ser[10100]: route[INBOUND]: (lookup_user($tu.uid, @ruri)) Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:540: class=GLOBAL Feb 13 14:36:42 rd ser[10100]: AVP["lang"]="en" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:580: track=TO class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:36:42 rd ser[10100]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:600: track=TO class=URI Feb 13 14:36:42 rd ser[10100]: INFO: No AVP present Feb 13 14:36:42 rd ser[10100]: route[INBOUND]: load_attrs($tu,$tu.uid) Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:540: class=GLOBAL Feb 13 14:36:42 rd ser[10100]: AVP["lang"]="en" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:550: track=FROM class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:560: track=TO class=DOMAIN Feb 13 14:36:42 rd ser[10100]: AVP["did"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["digest_realm"]="voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:570: track=FROM class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:580: track=TO class=USER Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:590: track=FROM class=URI Feb 13 14:36:42 rd ser[10100]: AVP["authuid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: AVP["uid"]="hellboy(a)voip.touk.pl" Feb 13 14:36:42 rd ser[10100]: INFO: avp.c:600: track=TO class=URI Feb 13 14:36:42 rd ser[10100]: INFO: No AVP present So generally lookup_contacts works only if it can find the uid in the uri class even if there are uid set in other classes it doesn' find the user. Another thing is that when invoked lookup_user("$t.uid","@ruri") the uid avp is put in the uri track. Maybe instead of lookup_contacts one should use registerd() function?? But what does it do? does it only check the location table or as well make an uri set?? Bests Tomasz > On Mon, 2007-02-12 at 22:13 +0100, tzieleniewski wrote: > > Thank You:) It help! > > But actually I had it done with the usage of lookup_user("$t.uid","@ruri"). > > does it make any difference from lookup_user("$tu.uid","@ruri") when I use $t.uid instead of $tu.uid? this is putting this avp to general class with the To track isn't it?? > > when I changed to lookup_user("Request-uri") it worked so generally works:) but not for lookup_user("$t.uid","@ruri"). > > Obviously there must be some difference... if it has helped. :-) You can > check what it has done using dump_attrs() call. It will output all AVPs > set to the syslog (must have debug >= 3 iirc). > > It should not go into general class, if not specified the user should be > the correct one... check the dump_attrs output and fill bug report at > tracker.iptel.org. (I guess it set the uri class and lookup_contacts > checks user class only) > > Michal > > > is there some order and priority in which the lookup_contacts() searches for uid?? (for instance first in user class and then I global??) > > Basically uri, user, domain, global is the order if you specify track > only AVP e.g. $t.uid so you can set widely accepted default value > (domain/global) and override it for some users and uris if you want. > > Some functions (like lookup_contacts) don't allow to set the avp name, > have to check the code what they expect. > > If you specify class too (e.g. $tu.uid) just the only one track/class is > searched. > > Michal > > > > > tomasz > > > > > The difference is that there is the forwarded request visible... > > > # > > > U 2007/02/12 20:35:16.241648 192.168.1.2:5060 -> 192.168.1.2:5060 > > > INVITE sip:hellboy@tezet.no-ip.org SIP/2.0. > > > Record-Route: <sip:192.168.1.2;ftag=1173592111;lr=on>. > > > Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKb946.ee8b4793.0. > > > Via: SIP/2.0/UDP > > > 192.168.1.2:7061;rport=7061;branch=z9hG4bK7AF94D00BA07C6380C5EC3434F4EB592. > > > From: tomix <sip:tomix@tezet.no-ip.org:7061>;tag=1173592111. > > > To: <sip:tomix@tezet.no-ip.org>. > > > Contact: <sip:tomix@192.168.1.2:7061>. > > > Call-ID: 0E6FCCFF-D661-0186-3392-977A4BBCDE86(a)192.168.1.2. > > > CSeq: 23808 INVITE. > > > Proxy-Authorization: Digest > > > username="tomix",realm="tezet.no-ip.org",nonce="45d0c2a0f928f22d790a5dfa17228b193d454c7e",response="0df094e8ea2808fa71d2fa28ccdfa8a4",uri="sip:tomix@tezet.no-ip.org",qop=auth,cnonce="67344073698582511BAA60061EBDA625",nc=00000001. > > > Max-Forwards: 16. > > > Content-Type: application/sdp. > > > User-Agent: X-Lite release 1105d. > > > Content-Length: 308. > > > > > > > > > If I understand it correctly you get this request being forwarded to the > > > contacs of tomix(a)tezet.no-ip.org and not hellboy@.... using usrloc > > > lookup (P-hint in the next INVITE). > > > > > > Then you must have set the TO / USER avp name "uid" based on the To > > > header instead of request-uri. Check the lookup_user function calls, it > > > should be > > > for REGISTER > > > lookup_user("To") or lookup_user("$tu.uid","@to.uri") > > > > > > and for other requests > > > lookup_user("Request-uri") or lookup_user("$tu.uid","@ruri") > > > > > > > > > If you want to do some originating services, you can use > > > lookup_user("From") or lookup_user("$fu.uid", "@from.uri") > > > > > > Michal > > > > > > > > > > > > On Mon, 2007-02-12 at 20:41 +0100, tzieleniewski wrote: > > > > Ok I did it. > > > > There is no difference. > > > > Just to make sure I attached the file. > > > > > > > > Because I left work I repeated the situation at home. I call to "myself" tomix(a)tezet.no-ip.org and I try to forward to hellboy(a)tezet.no-ip.org which is not registered. > > > > > > > > I can also send my ser.cfg if it might help. > > > > > > > > tomasz > > > > > > > > > I see. Please capture the network on linux cooked interface "any", so > > > > > even the request sent over loopback will be visible. > > > > > > > > > > Michal > > > > > > > > > > On Mon, 2007-02-12 at 16:53 +0100, tzieleniewski wrote: > > > > > > Yes but this ruri sip:hellboy@192.168.0.116:5060 is set after the invocation of lookup_contacts(). the function is invoked on the message which contains ruri changed by the forward_blind parameter. > > > > > > There is first "round" when processing of the first INVITE reaches the checking of the forward_blind parameter after which I invoke the attr2uri and just after this make the t_relay. then there is second "round" with the changed ruri. before lookup_contacts I see the ruri as mm(a)voip.touk.pl and after sip:hellboy@192.168.0.116:5060 which is in fact the location corresponding to sip:hellboy@voip.touk.pl. > > > > > > there is no message sending through the network but the message with the new ruri is processed which is visible in the log file I see it logged with the changed ruri?? > > > > > > > > > > > > So where is the problem?? > > > > > > Is it the problem of attr2uri? > > > > > > I tried it by using rewriteuri() and it gave me the same result. changed ruri but lookup_contacts() returns value corresponding to the first ruri. Maybe > > > > > > lookup_contacts() checks not the ruri?? > > > > > > > > > > > > > > > > > > > Yes, your request-uri is sip:hellboy@192.168.0.116:5060 which I think > > > > > > > will not be looked up in location (lookup_contact should fail), so it > > > > > > > does not rewrite the request-uri at all. > > > > > > > > > > > > > > Michal > > > > > > > > > > > > > > On Mon, 2007-02-12 at 16:05 +0100, tzieleniewski wrote: > > > > > > > > hmm > > > > > > > > it is strange because I can see that after attr2uri and t_relay() message again enters the main route block and goes through the whole processing but I can't see the message being send through the network. the recourd_route and via headers are being attached which is visible in the message send to unwanted (contained in the first invite - in this case myself because I try this by calling my self and setting forward_blind to another sip uri) destination: > > > > > > > > > > > > > > > > U 2007/02/12 16:05:12.199566 192.168.0.74:506 >

1 1

[Users] Send Call using authentication
by Jobson Andrade 13 Feb '07

13 Feb '07

Hi All, I need of send the calls for my ITSP using verification. Ex.: User = user_itsp Passwd = pass_itsp SIP Server = itsp.com.br I my configuration I made of this way, more not work. ... ##Obelisk if (uri=~"^sip:00[1-9][1-9][0-9]*@") { route(3); exit; } ... route[3]{ #sending route to DDI strip(2); rewritehostport("itsp.com.br:5060"); rewriteuser("user_itsp"); rewriteuserpass("pass_itsp"); route(1); exit; } ... How to made this in openser? I use the openser version 1.0.1 Thanks in advanced, Jobson Andrade Projetos & Desenvolvimento Obelisk - The Asterisk & VoIP Experts phone/fax: (11) 2164-4808 ext. 115 cell Phone: (11) 8175-9916 / 8271-0480 email: <mailto:jandrade@obelisknet.com.br> jandrade(a)obelisknet.com.br

1 0

[Users] Doubts about installation
by Javier Allende Astigarraga 13 Feb '07

13 Feb '07

i'm begining with openser , i've installed on ubuntu dapper but reading installation notes , always speak about openser-mysql-module, persistent, but i cant do apt.get install .... and if i seek i find files for http://packages.ubuntulinux.org/feisty/net/ only download and put con mysql/lib?? thanks for all any information about openserv config?? bye bye -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

2 1

[Serusers] call forwarding issue - attr_destination() unwanted behavior - ottendorf
by tzieleniewski 13 Feb '07

13 Feb '07

Hi!! I am trying to set up the blind call forwarding with the usage of attr_destination() function. I do the following I load the user attributes from user_attrs and uri_attrs table and then according to the loaded parameters I do: if ($tu.call_forward == "blind" && $tr.forward_blind) { xlog("L_INFO", " route[CALL_FORWARD]: fwd \n"); attr_destination("$tr.forward_blind"); xlog("L_INFO", " route[CALL_FORWARD]: route(FORWARD) \n"); route(FORWARD); xlog("L_INFO", " route[CALL_FORWARD]: drop \n"); drop; } after the ser.cfg logic goes through this part I don't see any change in the request uri?? Please point me what do I miss? here is my database contents: mysql> select * from user_attrs where uid like 'hellboy%' and name like 'call_%'; +----------------------+--------------+-------+------+-------+ | uid | name | value | type | flags | +----------------------+--------------+-------+------+-------+ | hellboy(a)voip.touk.pl | call_forward | blind | 2 | 1 | +----------------------+--------------+-------+------+-------+ 1 row in set (0.00 sec) mysql> select * from uri_attrs where username like 'hellboy%'; +----------+--------------+---------------+----------------------+------+-------+--------+ | username | did | name | value | type | flags | scheme | +----------+--------------+---------------+----------------------+------+-------+--------+ | hellboy | voip.touk.pl | forward_blind | sip:tzl@voip.touk.pl | 2 | 1 | sip | +----------+--------------+---------------+----------------------+------+-------+--------+ 1 row in set (0.00 sec) Cheers -tomasz

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[Users] Presence server -Confused about the Notify
by H Quintana 13 Feb '07

13 Feb '07

Hello, I'm a newbie with the SIMPLE extensions. This week I started using the presence server module (with the configuration suggested in the wiki and 2 X-lite's) The softphones are sending the SUBSCRIBE and the PUBLISH messages. The NOTIFYs are received for the Event presence.winfo with some XML content: <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="0" state="full"> <watcher-list resource="sip:10092011@a.b.c.d" package="presence"/> </watcherinfo> But, when I add each softphone as a Contact for the other there is no XML content for the Event presence, and the contact's status never changes even if I do multiple changes. I noticed that in the table xcap_xml there is no data, Should I put some information manually? or the information should be inserted when receiving the PUBLISH? Am I missing something? Any ideas? Thanks for your help, Humberto Quintana ____________________________________________________________________________________ Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121

2 1

[Users] several questions/concerns about branches
by Klaus Darilion 13 Feb '07

13 Feb '07

Hi! I hav the following questions about branch handling in openser 1.1.1: In route[1] I use the dispatcher to forward to the gateway 11.22.33.46. Thus, the DURI will be set. Further I use port 6060 to send calls to the GW. So far everything is ok. ... setting fr_timer to 2 seconds XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX route[1]: request's uri: $ru=sip:123456789@foobar.net route[1]: request's destination uri: $du=sip:11.22.33.46:5060 route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 route[1]: request's first branch: $br=<null> route[1]: request's all branches: $bR= route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net ... t_relay Then, if the GW does not reply within 2 seconds the failure route triggers: First question: Why is the DURI set to NULL now? ... entering failure route ERROR: no response from gateway or clients a1.net ... branches before ds_next_dst: $ru=sip:123456789@foobar.net, $du=, $bR= XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX failure_route[1]: request's uri: $ru=sip:123456789@foobar.net failure_route[1]: request's destination uri: $du= failure_route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 failure_route[1]: request's first branch: $br=<null> failure_route[1]: request's all branches: $bR= failure_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net ... ds_next_dst branches after ds_next_dst: $ru=sip:123456789@foobar.net, $du=, $bR=sip:123456789@foobar.net XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX failure_route[1]: request's uri: $ru=sip:123456789@foobar.net failure_route[1]: request's destination uri: $du= failure_route[1]: request's force_send_sock: $fs=udp:11.22.33.44:6060 failure_route[1]: request's first branch: $br=sip:123456789@foobar.net failure_route[1]: request's all branches: $bR=sip:123456789@foobar.net failure_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net, sip:979004369911 ... activating branch_route ... t_relay Second question: In the branch route the send_socket is reported as NULL. AFAIK it will be copied from branches[0] to the new branch during ds_next_dst. Also the request is sent from port 6060 - thus it looks like the $fs is wrong in branch_route. ==================================================== branch_route[1]: request's uri: $ru=sip:123456789@foobar.net branch_route[1]: request's destination uri: $du=sip:11.22.33.45:5060 branch_route[1]: request's force_send_sock: $fs=<null> branch_route[1]: request's first branch: $br=sip:123456789@foobar.net branch_route[1]: request's all branches: $bR=sip:123456789@foobar.net branch_route[1]: request's destination set: $ds=Contact: sip:123456789@foobar.net, sip:9790043699111 thanks klaus

5 8

[Serusers] branch parameter
by Kapil Dhawan 12 Feb '07

12 Feb '07

Whats the importance of branch parameter. I am using RTC/1.2 library and its not generating branch parameter. _________________________________________________________________ Tried the new MSN Messenger? Its cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in

5 7

Re: [Users] Corrupted 'To' after uac_restore_from
by Mahesh Paolini-Subramanya 12 Feb '07

12 Feb '07

Bogdan I'm sorry - I commited the cardinal cut/paste sin (copying stuff incorrectly :-( ) Anyhow, I've *correctly* cut/paste the INVITE and BYE headers below, and it looks to me like the route headers *are* being rewritten correctly. So, it doesn't look like that was the issue - unless you are referring to something else? cheers --- Original Message ----- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: mahesh(a)aptela.com Cc: users(a)openser.org Sent: Thursday, February 8, 2007 3:29:26 AM GMT-0600 Subject: Re: [Users] Corrupted 'To' after uac_restore_from Hi Mahesh, I suspect you have a client that does not preserve the Route/Record-Route parameters. According to RFC326, the parameters must not be altered at al, but I saw UAC doing lowercase on the param's value. Most probably this is your case also - the uac module use B64 encoding of the old uri in RR param, so this is case sensitive. Get the full trace of the call and check the original RR param (inserted by openser in INVITE) against the one in put in the BYE. regards, bogdan ---- CORRECTED MESSAGE----- I'm seeing this *very* weird error where uac_restore_from is ending up with a corrupted 'From' header. The weird thing is that it happens with only one end point (a Televantage SIP Trunk). With virtually *everything* else - and I am talking about tens of thousands of end points, spanning all sorts of sip devices, we are having *no* problems. The call is set up as follows -> the endpoint sends the INVITE, which is forwarded to a PSTN gateway The ACKs and the 200 OK all have their From/To headers appropriately rewritten The problem arises when the PSTN gateway initiates the BYE As you can see below, the 'To' header gets corrupted The *only* thing I can think of is that it has something to do with the structure of the Televantage tags (the '+' signs?). Any ideas? FYI, Openser 1.1.0 (latest CVS checkout), and modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","aptelavsf") --INVITE-- UAC->proxy INVITE sip:17038677006@69.25.47.136 SIP/2.0. Via: SIP/2.0/UDP 10.10.35.100;rport;branch=z9hG4bK+ec6a2a20436621337016546fa4dc9e87+10.10.35.100+1. Allow-Events: message-summary. Allow-Events: presence. Allow-Events: refer. Max-Forwards: 70. Call-ID: 56796C1E(a)10.10.35.100. From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>. CSeq: 132610708 INVITE. Expires: 180. Supported: replaces. Supported: 100rel. Contact: <sip:6462174165@10.10.35.100>. Content-Type: application/sdp. Content-Length: 242. . User-Agent: TeleVantage-UA/7.50.4817. . v=0. o=TeleVantage 10671 0 IN IP4 10.10.35.100. s=phone-call. c=IN IP4 10.10.35.100. t=0 0. m=audio 6036 RTP/AVP 0 18 101. a=rtpmap:0 pcmu/8000/1. a=rtpmap:18 g729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=ptime:20. ---INVITE--- proxy->gateway INVITE sip:17038677006@65.91.91.16:5060;transport=udp SIP/2.0. Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>. Via: SIP/2.0/UDP 69.25.47.136;branch=z9hG4bKf67d.2d31dbb.0. Via: SIP/2.0/UDP 10.10.35.100;rport=5060;branch=z9hG4bK+ec6a2a20436621337016546fa4dc9e87+10.10.35.100+1. Allow-Events: message-summary. Allow-Events: presence. Allow-Events: refer. Max-Forwards: 69. Remote-Party-ID: "Membersevice Queue" <sip:6462174165@flareon.aptela.com>;party=calling;id-type=subscriber;screen=yes;privacy=off. Supported: replaces. Call-ID: 56796C1E(a)10.10.35.100. From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>. CSeq: 132610708 INVITE. Expires: 180. Contact: <sip:6462174165@10.10.35.100:5060>. Content-Type: application/sdp. Content-Length: 262. User-Agent: TeleVantage-UA/7.50.4817. . v=0. o=TeleVantage 10671 0 IN IP4 10.10.35.100. s=phone-call. c=IN IP4 66.150.122.23. t=0 0. m=audio 59520 RTP/AVP 0 18 101. a=rtpmap:0 pcmu/8000/1. a=rtpmap:18 g729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=ptime:20. a=nortpproxy:yes . ---BYE---gateway->proxy BYE sip:6462174165@10.10.35.100:5060 SIP/2.0. Via: SIP/2.0/UDP 66.52.236.25:5060;branch=z9hG4bKd8orpb20eg2hais2c2g1sd0000g00.1. To: "Membersevice Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Call-ID: 56796C1E(a)10.10.35.100. CSeq: 1 BYE. Content-Length: 0. Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv>. Max-Forwards: 70. ---BYE--- proxy->UAC BYE sip:6462174165@10.10.35.100:5060 SIP/2.0. Record-Route: <sip:69.25.47.136;lr;ftag=1386496>. Via: SIP/2.0/UDP 69.25.47.136;branch=z9hG4bK7791.48e0e2f5.0. Via: SIP/2.0/UDP 66.52.236.25:5060;branch=z9hG4bKd8orpb20eg2hais2c2g1sd0000g00.1. To: "Membersevice Queue"<sip:user100.cru>6'(!.l asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Call-ID: 56796C1E(a)10.10.35.100. CSeq: 1 BYE. Content-Length: 0. Max-Forwards: 69. -- ******************************************* 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

[Users] MAJOR FIXUP: avp access in reply route
by Bogdan-Andrei Iancu 12 Feb '07

12 Feb '07

Hi everybody, The ability of accessing/processing AVPs in reply route was a long debated and wanted functionality. Finally, OpenSER 1.2.x fixes this. The new code allows two types of behaviours: - use per message AVPs in reply route - no special synchronization is needed, but there is no persistence outside reply route. - use per transaction AVPS in reply route (as in failure route). the above behaviours apply to the reply routes triggered by TM. You can control this via the new TM parameter "onreply_avp_mode": http://www.openser.org/docs/modules/1.2.x/tm.html#AEN303 in default reply route only per message AVPs will be available. regards, bogdan

3 3

[Users] invitation: SER User and Developer Meeting, collocated with the IETF, in Prague, March
by Jiri Kuthan 12 Feb '07

12 Feb '07

Ladies and gentlemen, boys and girls, be wholeheartly welcome to the SER user/developer meeting in Prague, March, colocated with the IETF meeting. For more information see http://www.iptel.org/ietf_meeting If interested in participation, just register yourself. Whereas this event is coming from the SER crowd, the content is going to be primarily technical and openser all in all draws greatly from SER. I believe this can be most useful for users(a)openser.org as well. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] ser log file
by zahra bahar 12 Feb '07

12 Feb '07

could we see the processes that ser has when receives each massage? is there any log file for ser in about this? --------------------------------- Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit.

1 0

[Users] DB tables for presence module
by Reinhold Buchinger 12 Feb '07

12 Feb '07

Hello, Can you explain me the difference between the information stored in the "watchers" table and "active-watchers" table? When is a watcher insert in which table? Thanks a lot! Best regards, Reinhold --

2 1

[Users] MEDIAPROXY - Cannot detect inactive rtp sessions, any other BYE fix method?
by Dan-Cristian Bogos 12 Feb '07

12 Feb '07

Hi Guys, I know that this is not the support channel of mediaproxy, but just hoping that somebody from ag-projects is monitoring this list as well. I found out that, in my case, the SIP-BYE issue is still persistent even by using mediaproxy . If I will stop sending rtp through mediaproxy (eg by killing my client from task manager, or loosing the internet), mediaproxy will still find the session active, like nothing happened and therefore the bill will be inaquarate. I am using the version 1.7.2, and as read in the change log, 1.8.0 didn't change anything on this chapter. Is there any other solution or workarround available? Thxs, Dan

1 1

[Serusers] Patch for MySQL 5.0.19 and above in SER 0.9.x
by sip 12 Feb '07

12 Feb '07

I've attached a patch file for people to use (it really only adds 3 lines) to patch the my_con.c file in the mysql module for SER 0.9.x I've tested it for a week with inactivity, restarts of mysql, etc, and it holds up rather well, but of course I would be remiss if I didn't add a 'use at your own risk' disclaimer. ;) The basic problem is that, after MySQL 5.0.3, they changed the default behaviour of the mysql connection socket to NOT reconnect in the case of a timeout or a restart of the server. If you had SER up and running for a while, or if there were a glitch with your mysql, the connection would time out/disconnect, and SER would no longer be able to hit the database (requiring a restart of SER). The MySQL coders added an option to enable the reconnect ability of the mysql connection, but it didn't work until 5.0.19. SO... if you're using mysql 5.0.19 or above with SER 0.9.x (there may only be 3 of us doing this ;) ), here's a patch file for you to apply to my_con.c in your modules/mysql directory. I'd recommend backing up your my_con.c file before patching. N.

1 0

[Users] ?opsr? - setup
by Michel Bensoussan 12 Feb '07

12 Feb '07

Hello I have some setup problems. I compiled and install as following: /prompt$ make mode=debug all prompt$ su prompt# make install /To run openSer I tried: /prompt# /usr/local/sbin/openser / and /prompt# /usr/local/sbin/openser D E / in both case there is the following output: / 127.0.0.1 [127.0.0.1]:5060 192.168.1.86 [192.168.1.86]:5060 192.169.4.86 [192.169.4.86]:5060 192.168.13.86 [192.168.13.86]:5060 127.0.0.1 [127.0.0.1]:5060 192.168.1.86 [192.168.1.86]:5060 192.169.4.86 [192.169.4.86]:5060 192.168.13.86 [192.168.13.86]:5060 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.86 [192.168.1.86]:5060 udp: 192.169.4.86 [192.169.4.86]:5060 udp: 192.168.13.86 [192.168.13.86]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.86 [192.168.1.86]:5060 tcp: 192.169.4.86 [192.169.4.86]:5060 tcp: 192.168.13.86 [192.168.13.86]:5060 Aliases: tcp: michel:5060 tcp: michel.local:5060 tcp: localhost:5060 udp: michel:5060 udp: michel.local:5060 udp: localhost:5060 //prompt# /Attached my config file ( /etc/openser/openser.cfg). On this config file, I set listen=192.168.13.86 so I expected the openSER to listen only 192.168.13.86. Actually I don't understand from where the application takes the openser.cfg file? I removed /etc/openser/openser.cfg but the application is still running. / /Regards, Michel. / / # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode */ fork=no log_stderror=yes /**/ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.13.86 port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

2 1

Re: [Users] A better way to count the length of a variable
by Tavis P 12 Feb '07

12 Feb '07

Ah efficient length counting, you're only a version away... Thanks Brandon Brandon Armstead wrote: > Tavis P wrote: >> I'm trying to find a fast way to count the length of a variable (and >> detect when the length exceeds a certain value) >> All i've come up with so far is a regex, does anyone know of a >> faster/better way to do this? >> >> >> Heres my current approach: >> !avp_check("$ci", "re/^.{1,128}$/g") >> >> >> tavis >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > Hello Tavis, > > Currently that is the only method available, however keep your eye > out for the release of OpenSER version 1.2.0, it was recently froze in > CVS and hopefully will be ready pretty soon, however here are some of > the methods that will be available: > http://openser.org/dokuwiki/doku.php/transformations:devel. > > Regards, > Brandon > >

1 0

[Users] A better way to count the length of a variable
by Tavis P 12 Feb '07

12 Feb '07

I'm trying to find a fast way to count the length of a variable (and detect when the length exceeds a certain value) All i've come up with so far is a regex, does anyone know of a faster/better way to do this? Heres my current approach: !avp_check("$ci", "re/^.{1,128}$/g") tavis

1 0

[Serusers] Problems with Rtpproxy traffic balancing in SER 0.10.x
by Angelo Taddeo 12 Feb '07

12 Feb '07

I'm using ser 0.10.x ( CVS version). About rtpproxy traffic balancing, I don't know where I have to put rtpproxy IP address and where I have to set $node variable value. How does SER obtain rtpproxy IP address ? What the relation between rtpproxy IP address and N0, N1.....etc ? What parameters have I to put in force_rtp_proxy() function? Only rtpproxy IP address? In my framework happens : when SER receives INVITE Request and 200 OK it rewrites the body of SIP message (SDP). The problem is that it doesn't put in connection information (c) rtpproxy IP address; it puts its IP address. So clients send their RTP packet to SER instead to rtpproxy. How I can force SER to rewrite correctly SDP so that clients receive rtpproxy IP address in INVITE and 200 OK SIP message? Thanks in advance Alekos --------------------------------- L'email della prossima generazione? Puoi averla con la nuova Yahoo! Mail

1 0

[Serusers] register with ser
by zahra bahar 12 Feb '07

12 Feb '07

Hi. I installed SER.when I want to register with xlite in ser.xlite phone sends INVITE to ser.but ser doesn't send ok or any reply. what is the wrong. wen I run ser I have: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.10.10.138 [10.10.10.138]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.10.10.138 [10.10.10.138]:5060 Aliases: tcp: Shahab2:5060 tcp: Shahab2.Sarv:5060 tcp: localhost:5060 udp: Shahab2:5060 udp: Shahab2.Sarv:5060 udp: localhost:5060 WARNING: no fork mode and more than one listen address found(will use only thethe first one) stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 112640 0(0) INFO: udp_init: SO_RCVBUF is finally 225280 0(0)WARNING: using only the first listen address (no fork) 2(6248) INFO: fifo process starting: 6248 2(6248) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(6248) WARNING: no fifo_db_url given - fifo DB commands disabled! --------------------------------- Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games.

3 2

RE: [Serusers] register with ser
by Su Kerry 10 Feb '07

10 Feb '07

1 0

[Users] NAT problem for PRACK replies
by Jeff Williams 09 Feb '07

09 Feb '07

Hi, I seem to have a problem replies to PRACKs from NATed UAs. The UA is behind the NAT on a private IP listening on port 5060. This port gets mapped to 1350 on the external IP by the NAT device. The INVITE goes out fine and gets the contact updated for NAT by: if (client_nat_test("3")) { setflag(7); force_rport(); fix_contact(); }; This then receives a 183 reply to which it sends a PRACK. However, the 200 response to the PRACK gets sent to port 5060 on the external IP => which of course doesn't get to the UA. Is there something I'm missing in my config? I have attached a trace of the call setup. Jeff No. Time Source Destination Protocol Info 19424 6.838985 203.24.54.17 202.74.164.44 SIP/SDP Request: INVITE sip:0893680949@proxy.sipone.com;user=phone, with session description Frame 19424 (941 bytes on wire, 941 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:0893680949@proxy.sipone.com;user=phone SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Max-Forwards: 70 Contact: Jeff <sip:610089001@192.168.10.30:5060;user=phone;transport=udp> User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 288 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 610089001 147590 147590 IN IP4 192.168.10.30 Session Name (s): Cisco 7912 SIP Call Connection Information (c): IN IP4 192.168.10.30 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 16384 RTP/AVP 0 18 8 101 Media Attribute (a): rtpmap:0 PCMU/8000/1 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:8 PCMA/8000/1 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 19428 6.842541 202.74.164.44 203.24.54.17 SIP Status: 100 trying -- your call is important to us Frame 19428 (655 bytes on wire, 655 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: 1350 (1350) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK26f185af4ccc0d42;rport=1350;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:0893680949@proxy.sipone.com;user=phone out_uri=sip:61893680949@202.74.164.72:5060;user=phone via_cnt==1" No. Time Source Destination Protocol Info 19429 6.842579 202.74.164.44 202.74.164.72 SIP/SDP Request: INVITE sip:61893680949@202.74.164.72:5060;user=phone, with session description Frame 19429 (1148 bytes on wire, 1148 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:61893680949@202.74.164.72:5060;user=phone SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone> Call-ID: 3478456911(a)192.168.10.30 CSeq: 1 INVITE Max-Forwards: 69 Contact: Jeff <sip:610089001@203.24.54.17:1350;user=phone;transport=udp> User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 288 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 610089001 147590 147590 IN IP4 192.168.10.30 Session Name (s): Cisco 7912 SIP Call Connection Information (c): IN IP4 202.74.164.44 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 63054 RTP/AVP 0 18 8 101 Media Attribute (a): rtpmap:0 PCMU/8000/1 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:8 PCMA/8000/1 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 19450 6.860058 202.74.164.72 202.74.164.44 SIP Status: 100 Trying Frame 19450 (445 bytes on wire, 445 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header From: "Jeff" <sip:0892209080@sipone.com:5060>;tag=1166274291 To: <sip:0893680949@202.74.164.72:5060;user=phone>;tag=669b638bbf9e0903fd8a2a66 CSeq: 1 INVITE Call-ID: 3478456911(a)192.168.10.30 Via: SIP/2.0/UDP 202.74.164.44:5060;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 Content-Length: 0 No. Time Source Destination Protocol Info 21001 8.339800 202.74.164.72 202.74.164.44 SIP/SDP Status: 183 Session Progress, with session description Frame 21001 (1064 bytes on wire, 1064 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK7a7e.3cf74a67.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:07 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Require: 100rel RSeq: 3067 Contact: <sip:610089001@202.74.164.72:5060;user=phone> Record-Route: <sip:202.74.164.72:5060;lr> Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 214 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent 4912 6655 IN IP4 202.74.164.72 Session Name (s): SIP Call Connection Information (c): IN IP4 202.74.164.72 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 27182 RTP/AVP 0 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 21003 8.341340 202.74.164.44 203.24.54.17 SIP/SDP Status: 183 Session Progress, with session description Frame 21003 (1016 bytes on wire, 1016 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: 1350 (1350) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;rport=1350;received=203.24.54.17;branch=z9hG4bK26f185af4ccc0d42 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:07 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Require: 100rel RSeq: 3067 Contact: <sip:610089001@202.74.164.72:5060;user=phone> Record-Route: <sip:202.74.164.72:5060;lr> Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU-> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 214 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent 4912 6655 IN IP4 202.74.164.72 Session Name (s): SIP Call Connection Information (c): IN IP4 202.74.164.44 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 63054 RTP/AVP 0 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 No. Time Source Destination Protocol Info 21044 8.397481 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21044 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21045 8.397734 202.74.164.44 202.74.164.72 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21045 (725 bytes on wire, 725 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291> Route: <sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.215a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 69 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21060 8.419413 202.74.164.72 202.74.164.44 SIP Status: 200 OK Frame 21060 (489 bytes on wire, 489 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.215a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:08 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 21061 8.419532 202.74.164.44 203.24.54.17 SIP Status: 200 OK Frame 21061 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:08 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 21070 8.429213 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 21070 (469 bytes on wire, 469 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 21611 8.897497 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 21611 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 21612 8.897734 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 21612 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 21625 8.909606 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 21625 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 22692 9.898092 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 22692 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 22693 9.898343 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 22693 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11078 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 22702 9.912093 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 22702 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 24851 11.899557 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 24851 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 24853 11.899808 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 24853 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11081 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 24866 11.911720 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 24866 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 27365 13.900358 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 27365 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 27366 13.900630 202.74.164.44 202.74.164.72 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 27366 (725 bytes on wire, 725 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 202.74.164.72 (202.74.164.72) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Record-Route: <sip:202.74.164.44;lr=on;ftag=1166274291> Route: <sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.315a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 69 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 27426 13.938284 202.74.164.72 202.74.164.44 SIP Status: 200 OK Frame 27426 (489 bytes on wire, 489 bytes captured) Ethernet II, Src: 00:12:44:c0:cc:1b, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 202.74.164.72 (202.74.164.72), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 202.74.164.44;branch=z9hG4bK4a7e.315a51c5.0 Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: "Jeff" <sip:0892209080@sipone.com>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:14 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 27427 13.938422 202.74.164.44 203.24.54.17 SIP Status: 200 OK Frame 27427 (441 bytes on wire, 441 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;received=203.24.54.17;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Date: Fri, 09 Feb 2007 00:15:14 GMT Call-ID: 3478456911(a)192.168.10.30 Server: Cisco-SIPGateway/IOS-12.x CSeq: 2 PRACK Content-Length: 0 No. Time Source Destination Protocol Info 27438 13.947350 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 27438 (469 bytes on wire, 469 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol No. Time Source Destination Protocol Info 30349 15.901399 203.24.54.17 202.74.164.44 SIP Request: PRACK sip:610089001@202.74.164.72 Frame 30349 (711 bytes on wire, 711 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) User Datagram Protocol, Src Port: 1350 (1350), Dst Port: sip (5060) Session Initiation Protocol Request-Line: PRACK sip:610089001@202.74.164.72 SIP/2.0 Message Header Route: <sip:202.74.164.44;lr=on;ftag=1166274291;nat=yes;vsf=AAAAAAYJCQIKCQkACXAwAQYIFkAWRxMAA2UuY29tO3VzZXI9cGhvbmU->,<sip:202.74.164.72:5060;lr> Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK RAck: 3067 1 INVITE Max-Forwards: 70 User-Agent: Cisco-CP7912/8.0.1-060412A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Content-Length: 0 No. Time Source Destination Protocol Info 30350 15.901638 202.74.164.44 203.24.54.17 SIP Status: 500 I'm terribly sorry, server error occurred (1/SL) Frame 30350 (649 bytes on wire, 649 bytes captured) Ethernet II, Src: 00:07:e9:11:47:25, Dst: 00:12:44:c0:cc:1b Internet Protocol, Src Addr: 202.74.164.44 (202.74.164.44), Dst Addr: 203.24.54.17 (203.24.54.17) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) Message Header Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKeccd468d60f015ed;received=203.24.54.17 From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1166274291 To: <sip:0893680949@proxy.sipone.com;user=phone>;tag=669b638bbf9e0903fd8a2a66 Call-ID: 3478456911(a)192.168.10.30 CSeq: 2 PRACK Server: OpenSer (1.1.1-tls (i386/linux)) Content-Length: 0 Warning: 392 202.74.164.44:5060 "Noisy feedback tells: pid=11082 req_src_ip=203.24.54.17 req_src_port=1350 in_uri=sip:610089001@202.74.164.72 out_uri=sip:610089001@202.74.164.72 via_cnt==1" No. Time Source Destination Protocol Info 30359 15.913057 203.24.54.17 202.74.164.44 ICMP Destination unreachable (Port unreachable) Frame 30359 (590 bytes on wire, 590 bytes captured) Ethernet II, Src: 00:0a:f3:b8:0c:08, Dst: 00:07:e9:11:47:25 Internet Protocol, Src Addr: 203.24.54.17 (203.24.54.17), Dst Addr: 202.74.164.44 (202.74.164.44) Internet Control Message Protocol

4 4

[Users] avpops on reply_route
by Klaus Darilion 09 Feb '07

09 Feb '07

Hi! Once again I want to discuss the problem of avpops in reply_routes (e.g. http://www.openser.org/pipermail/users/2005-October/001281.html) I can bypass the syntax parser by putting the AVP-commands into a normal route block which is executed from the reply_route. As far as I understand this may cause problems as the AVP-ops are done without locking. But what may happen in worst case? Bad AVPs or a crash? thanks klaus btw: is there any other way to handle data from a reply_route ?

2 1

Re: [Serusers] failed to load mysql.so module
by Josef.Bajada＠go.com.mt 09 Feb '07

09 Feb '07

3 5

[Users] switch - pseudo vars
by Cseke Tamas 09 Feb '07

09 Feb '07

Hello, I'm using openser 1.1.0 I'd like to make a dinamic routing, the plan is radius set an avp, which contains the routeid, but my config is bad, and i don't know what can be the problem route($avp(i:20)); don't work so i tryied this way: switch($avp(i:20)) { case 0: route(0); break; case 1: route(1); break; case 2: route(2); break; } but this is bad too, can anybody give me an advice? the problem is in the line "switch($avp(i:20)) {" i guess, but switch can accept pseudo vars, so i don't kow what the problem caould be? Thanks any help, Tamas

3 5

[Users] using multiple DNS servers
by Will Quan 09 Feb '07

09 Feb '07

I have a primary and secondary dns server setup in resolv.conf, with a timeout option of 1 second.The idea is when the primary fails, my secondary server picks up the workload.When openser is running, and primary dns is down, I am seeing several re-transmissions of 200 OK (invite) to my originator *after* openser has finished processing ACK. I was thinking the final responses would stop reTX after the ACK was t_relayed. Have you seen such behavior? And better yet- Is there a solution for this?Thanks, Will

4 3

[Serusers] Radius accounting - User-Name parameter in the Start Acct-Status-Type
by tzieleniewski 09 Feb '07

09 Feb '07

Hi! Is it possible to configure acc_radius in such a way that it will send a User-Name parameter in the Accounting Packet because now it doesn't: rad_recv: Accounting-Request packet from host 127.0.0.1 port 32844, id=246, length=474 SER-Attr = "" Acct-Session-Id = "6012E956-D8D3-2367-979B-753579FE59FF(a)192.168.1.2" Sip-To-Tag = "572241288" SER-From = "tomix <sip:tomix@tezet.no-ip.org:7060>;tag=2020612254" SER-Flags = 8 SER-Original-Request-ID = "sip:hellboy@tezet.no-ip.org" Sip-Method = "INVITE" Sip-Cseq = 20886 Sip-Translated-Request-ID = "sip:hellboy@192.168.1.10:5060" Sip-Source-IP-Address = 192.168.1.2 Sip-From-Tag = "2020612254" SER-To = "<sip:hellboy@tezet.no-ip.org>;tag=572241288" SER-Digest-Username = "tomix" SER-Request-Timestamp = 1171023334 Calling-Station-Id = "sip:tomix@tezet.no-ip.org:7060" Sip-Source-Port = 7060 SER-Digest-Realm = "tezet.no-ip.org" Sip-Response-Code = 200 Called-Station-Id = "sip:hellboy@tezet.no-ip.org" SER-Response-Timestamp = 1171023341 Acct-Status-Type = Start Service-Type = IAPP-Register NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Cheers Tomasz

1 0

[Serusers] SER 0.10.99 (Carpo) - NAT call - long ringing after hang up
by Theodor Meier 09 Feb '07

09 Feb '07

hello, using my fritzbox fon with an account from Carpo, ser 0.10.99, to call a cisco 7960 behind the same fritzbox fon, that uses as well an Carpo account, an hang up, the called cisco 7960 continues ringing for a minute. (fritzbox - ser - (same) fritzbox - 7960) Vice versa the same. (7960 - fritzbox - ser - fritzbox) Not so when I use the 7960 with a sipgate account (7960 - sipgate - telefonica gateway - Carpo ser - fritzbox) to call the fritzbox. Is this a ser problem? Perhaps a problem, when the call-phone and the called-phone use the same IP? Or is this a configuration problem of ser? What can I debug to find where the problem is? ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

5 6

[Users] RTP fixes for mediaproxy
by Jeff Williams 09 Feb '07

09 Feb '07

Hi, I was having a few issues with mediaproxy when it came to re-invites and call setup where the SDP does not get sent with the first INVITE. I noticed this was due a couple couple of assumptions in mediaproxy which don't always hold: 1) That SDP is always set up first by the caller (i.e. in the INVITE) then updated by the called party (i.e. in the 200). This is not the case. The first SDP can be sent in the 200 response by the called party and the caller send their SDP in the ACK. 2) That RTP streams don't change once set up. This is wrong for reINVITEs. This would cause problems when a user agent sent a reINVITE to redirect RTP to on hold music for example - the RTP change would be ignored. I think this had an impact on some fax setups as well, but I haven't tested it. I found a patch to fix this by storing the streams in a hash by media type, but that is a limiting assumption also (maybe I want 2 audi streams?). The attached patch fixes this by changing the following: 1) This pathc allows a mediaproxy RTPSession to be set up by either the caller or called party. A 'caller' argument is passed to the RTPSession functions telling it if the caller or called party is doing the setup or update. 2) The official mediaproxy does not even look at SDP details for streams which are already set up. This patch lets every SDP packet update all of the RTP streams for that party (either caller or called). If you have issues with one-way-audio with media proxy after reINVITEs, have calls the first SDP from the called party or are just plain adventurous, please give it a go. It has been used in a production environment for a bit over a month now. The patch was made against mediaproxy 1.7.2, but will work against 1.8.0 as well since none of the rtphander.py code changed. Regards, Jeff

2 2

[Serusers] failed to load mysql.so module
by Josef.Bajada＠go.com.mt 08 Feb '07

08 Feb '07

Hi, I am having this problem with the mysql module on ser. 0(31212) ERROR: load_module: could not open module </opt/ser/modules/mysql/mysql.so>: /opt/ser/modules/mysql/mysql.so: undefined symbol: log 0(31212) parse error (30,13-14): failed to load module ERROR: bad config file (1 errors) I can confirm that mysql.so exists in that directory. I have mysql 5.0.27 installed. I have already tried to clean and make all from scratch but the problem remained. Any ideas how to solve this? Thanks a lot, Josef

3 2

[Users] set RURI with dispatcher module
by Klaus Darilion 08 Feb '07

08 Feb '07

Hi! I want to use the dispatcher module to use load-balancing/failover to gateways. Thus, I also want to change the domainpart of the RURI to be the IP address of the gateway - but dispatcher only sets the DURI. Is there an easy way to set the RURI/domain too or do I have to use: avp_write("$rU","$avp(i:666)"); ds_select_dst(...); avp_pushto("$ru","$du"); avp_pushto("$ru/username","$avp(i:666)"); Is there a more easy way? btw - I think above workaround would not work in failure_route as ds_next_dst adds a new branch but avp_pushto works against branches[0] - correct? thanks klaus

3 4

[Users] SIP Proxy - CAC
by Michel Bensoussan 08 Feb '07

08 Feb '07

Hello, I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization, for example according to codec, and allow or disallow new sessions, depending on resources. I'm thinking using OpenSER for this purpose. On a first stage I want to install a SIP Proxy on the AP (embedded with OS Linux) and on a second stage adding the CAC feature. There is many modules in OpenSER and I'm not sure which one I should use for a SIP Proxy. I'm learning about SIP but my knowledge is limited (till last week I never heard about it). Thank you for advising. Regards, Michel.

2 3

[Serusers] 0.9.x mysql.so: undefined symbol: log error
by Steve Blair 08 Feb '07

08 Feb '07

Hi: I keep getting the following error on a 0.9.x system. The latest 0.9.x version of SER was just download and built on this server but I continue to get this same error. I've check the mysql Makefile and it seems to be correct. What else could cause this error? ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Thanks,Steve

4 8

[Serusers] sequential hunting
by Steve Blair 08 Feb '07

08 Feb '07

Hello: I would like to implement sequential hunting. Hunting is when a call arrives for a given subscriber, no one answers the original number so the proxy populates the username portion of the r-uri with another number (from the database) and tries again. This process continues until all numbers in the database are exhausted. I have the "failover" working but the r-uri field is not updated on any interation though the failure_route. Does anyone have any ideas how to make the following work? Thanks,Steve # hunting (serial forking) if (avp_db_load("$ruri/username", "s:allow_hunt")) { if (avp_check("s:allow_hunt", "eq/y/i")) { avp_delete("s:allow_hunt"); xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Enabled for <%ru>\n"); avp_db_load("$ruri/username", "s:huntlist/sam"); xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Call Hunting Started to <%ru>\n"); t_on_failure("6"); t_relay(); break; }; }; # Iterate through extension list for subscribers with call hunting enabled failure_route[6] { xlog("L_INFO", "\n[SER]: [%Tf] [%ci] Failure Block #6: HUNTING for <%ru> from <%fu> at <%is>\n"); if (avp_pushto("$ruri/username", "s:huntlist/g")) { append_branch(); avp_delete("s:huntlist"); t_on_failure("6"); t_relay(); }; }

5 12

[Users] RADIUS acc ok, RADIUS auth not
by Luca Corti 08 Feb '07

08 Feb '07

Hello, I'm having a hard time getting RADIUS auth to work at all. I've reached the point where I am debugging the network traffic with tshark. I can see RADIUS packets generated from the openser box for accounting, but no RADIUS authentication packets are sent. Also, it seems no response comes back from the RADIUS server even for accounting requests. radclient against my radius server returns me an Access-Reject at least and the tshark dump contains auth request and response. OpenSER 1.1.1 tells me: Feb 6 17:27:04 openser openser[22596]: rc_send_server: no reply from RADIUS server I suspect the issues lie in the radiusclient-ng layer. I'm using radiusclient-ng from Debian Etch (0.5.3-2), but I see many branches in CVS, like RELENG_0_5_5 which seems newer. thanks

2 6

[Users] uac_replace_from() not work for Linksys ATA
by Howard Tang 08 Feb '07

08 Feb '07

HI All, I have a problem with uac_replace_from() function not working for Linksys Unit. This is what i have in the route[1], if (search("From:.*<sip:900*")) { uac_replace_from("anonymous","sip:anonymous@x.x.x.x"); } I have 0900xx as internal sip account, then I have to remove the 0900xx callerid before i send the call to PSTN gateway. Because 0900xx is not a valid number, Mobile phone providers block the call because of that, as a result, I need to replace the callerid to anonymous. I have tested with many other ATA and X-Lite without problem. The problem only ocurrs when i use Linksys ata. I have monitored the SIP message, the first request will change to anonymous then back to the original username. Anyone have an idea on how i can fix this ? -- Howard Tang ICQ : 259083 MSN : howard615(a)hotmail.com

2 7

[Serusers] Digest Authentication in Ottendorf server
by raj gopal 08 Feb '07

08 Feb '07

Hi, I can not use www_challenge(realm,qop)function in ser.cfg file even though auth module is loaded. The readme file has documentation for it. When I use the following in config file and start SER, the following error is thrown if (!radius_www_authorize("")) { www_challenge("", "1"); // parse error break; }; parse error (197,49-50): unknown command, missing loadmodule? I am using Ottendorf release. How to do digest authentication with new release??** <http://ftp.iptel.org/pub/ser/daily-snapshots/unstable/> -Venkat

2 1

[Users] Angle bracket in Route header coming from Contact and strict router
by Marcello Lupo 08 Feb '07

08 Feb '07

Hi to all, little question about populating the Route Header field. I'm dealing with 2 CPE of different vendors (Patton and Netsynt) and an Openser in the middle. Patton in their SIP is not using angle bracket on SIP From,To and Contact headers. Netsynt is using strict routing method to build BYE messages. This is causing that when Netsynt is creating the BYE it is putting the proxy IP in the R-URI and the URI of Contact received from previuos packets from patton in the Route header to let it be processed by the proxy. In this way the Route header field constructed is without the <> enclosing the uri. Netsynt keep it from previous Contact without any manipulation on it. This is causing an error on the proxy parsing the name-addr field of the Route header field and the loose_route() fail. Seems that it is mandatory for Openser to get angle bracket in Route header fields. Now in the section 20 of RFC3261 is said that it is not a MUST that From,To and Contact header have to be enclosed in angle bracket. So i'm searching in the RFC where is said that if the Contact is without the angle bracket the UAC have to check if the Contact have the <> and if it don't have the <> it have to insert it before to out it on the Route (i searched the RFC2543 too). can someone give me clarification on this issue? Where is written the Route header field is mandatory to have <> enclosing the URI. I'm searching all of this because i have to give a good reason to each vendor (Netsynt and Patton) to modify their sip stack for this issue. I don't know who of the 2 should fix the problem. If Patton to add <> in each request or Netsynt to fix the Route field population. Thanks, Bye, Marcello

2 1

RE: [Serusers] xlite and ser
by Su Kerry 08 Feb '07

08 Feb '07

1 0

[Serusers] xlite and ser
by zahra bahar 08 Feb '07

08 Feb '07

Hi all. could any one send me all the things that I should do for having only a call between two xlite phones with ser server? I have installed ser and xlite on 10.10.10.138 and another xlite phone on 10.10.10.195. the xlite phone on 10.10.10.195 sends register massage to 10.10.10.138 but xlite phone that is on 10.10.10.138 doesn't send massage to ser. I didn't change any thing in ser only fork=no,log-stderror=yes.children=4. and I don't user sql. thanks. --------------------------------- Access over 1 million songs - Yahoo! Music Unlimited.

1 0

[Users] Corrupted 'To' after uac_restore_from
by Mahesh Paolini-Subramanya 08 Feb '07

08 Feb '07

I'm seeing this *very* weird error where uac_restore_from is ending up with a corrupted 'From' header. The weird thing is that it happens with only one end point (a Televantage SIP Trunk). With virtually *everything* else - and I am talking about tens of thousands of end points, spanning all sorts of sip devices, we are having *no* problems. The call is set up as follows -> the endpoint sends the INVITE, which is forwarded to a PSTN gateway The ACKs and the 200 OK all have their From/To headers appropriately rewritten The problem arises when the PSTN gateway initiates the BYE As you can see below, the 'To' header gets corrupted The *only* thing I can think of is that it has something to do with the structure of the Televantage tags (the '+' signs?). Any ideas? FYI, Openser 1.1.0 (latest CVS checkout), and modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","aptelavsf") ---INVITE--- endpoint->proxy From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8d200003+32cc4d0e. To: <sip:17038677006@69.25.47.136>;tag=277e3aa2408f2ad73a234f5bbb99133b.c13b. ---INVITE--- proxy->gateway From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136> Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> --- 183 Session Progress--- gateway->proxy From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496. Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---183 Session Progress--- proxy->endpoint From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 ---200 OK--- gateway->proxy From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---200 OK--- proxy->endpoint From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---BYE--- gateway->proxy To: "Membersevice Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---BYE--- proxy->endpoint To: "Membersevice Queue"<sip:user100.cru>6'(!.l asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. -- ******************************************* 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 1

[Users] qop="auth" with uac_auth
by Andreas Heise 08 Feb '07

08 Feb '07

Hello, since a lot of providers has change there authentication to qop="auth" the uac_auth(); function of the uac module can't use anymore. by google I found a lot of requests for qop with uac_auth and also the official feature request 1345887. Is a target date known for a solution, maybe it's possible with 1.2.0? Feature Request [ 1345887 ] Implement qop functionality in uac module p.s.: I know it's on the road map, but it is needed so often...... thanks. Andreas

4 10

[Users] Reg:NAT Routing
by venkid 08 Feb '07

08 Feb '07

hai, can any body help me to test audio call between two different private IPs. am using X-lite as client .am not able to get voice between two private ips and able to get voice with public IP .i have run media proxy saparatly. please find attached my openser.cfg file too. Thanks and Regards venkatesh.d debug=9 fork=yes log_stderror=yes listen=192.168.0.25 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openser@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/presence.so" modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://openser:openser@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("presence", "db_url","mysql://openser:openser@localhost/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'a') modparam("presence", "totag_avpid", 987) modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "force_active", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE log("setting nat as yes *****************"); record_route_preset("192.168.0.25:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { log("setting falg 6 here **********"); setflag(6); use_media_proxy(); }; }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); exit; }; if (method=="ACK") { route(1); exit; } else if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { # if (method="INVITE" || method=="ACK") { # end_media_session(); # }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

1 0

Re: [Serusers] Problemswith compiling ser 0.10.99
by Dmitry Isakbayev 07 Feb '07

07 Feb '07

The OSP toolkit project has been moved from sipfoundry to sourceforge - http://sourceforge.net/projects/osp-toolkit Regards, Dmitry > Date: Mon, 08 Jan 2007 10:10:47 -0500 > From: Dmitry Isakbayev <dmitry(a)transnexus.com> > Subject: Re: [Serusers] Problemswith compiling ser 0.10.99 > To: "tzieleniewski" <tzieleniewski(a)o2.pl> > Cc: "'Support@Transnexus. Com'" <support(a)transnexus.com>, > serusers(a)lists.iptel.org > Message-ID: <45A25EF7.7070502(a)transnexus.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi, > > The OSP toolkit project is being moved from sipfoundry to sourceforge. > It should be available shortly. Meanwhile, the latest snapshot of the > toolkit can be downloaded from > > http://www.transnexus.com/OSP%20Toolkit/OSP%20Toolkit.htm > > Regards, > Dmitry > > Date: Mon, 08 Jan 2007 11:25:24 +0100 > > From: "tzieleniewski" <tzieleniewski(a)o2.pl> > > Subject: [Serusers] Problemswith compiling ser 0.10.99 > > To: serusers(a)lists.iptel.org > > Message-ID: <54c54dce.20e2e287.45a21c14.f0624(a)o2.pl> > > Content-Type: text/plain; charset="utf-8" > > > > Hi All!! > > I am trying to compile the lattest cvs version of ser but I get errors > > which I cann't handle myself so i kindly ask for your help:) > > What I want to achive basically is the ser instance with presence > > support. > > I am compiling ser in the following way: > > make group_include="standard standard-dep stable" all > > I read that osp requires osptoolkit i tried to get it from sipfoundry > > but i couldn't find any source of that. > > I also cann't fight problems with tls i didn't includethe experimental > > modules(tls) but still I get some errors.

2 1

[Serusers] SIP Proxy - CAC
by Michel Bensoussan 07 Feb '07

07 Feb '07

Hello, I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization, for example according to codec, and allow or disallow new sessions, depending on resources. I'm thinking using SER (or maybe OpenSER) for this purpose. On a first stage I want to install a SIP Proxy on the AP (embedded with OS Linux) and on a second stage adding the CAC feature. There is many modules in SER and I'm not sure which one I should use for a SIP Proxy. I'm learning about SIP but my knowledge is limited (till last week I never heard about it). Thank you for advising. Regards, Michel.

1 0

[Serusers] Newbie ser user, moving from asterisk
by Antonio José dos Santos Brandão 07 Feb '07

07 Feb '07

Hi, I'm starting with ser and would like to know if it's possible no make the following: Today i use asterisk with sip users registered to it and a DeadAgi (a2billing) to manage calls and prepaid/postpaid accounts. Works fine. Since i have too many features and customizations i can't abone it. The problem is that many users are behind nat and have to canreinvite=no, forcing the rtp media to passthru asterisk. I would like to implement media_proxy in this scenario to balance the rtp's bandwith with another hosts. If there were some kind of media_proxy for asterisk would be great, but there isn't. So, i wonder: can I put ser in this, after or before asterisk, making possible to user media_proxy without change so much in asterisk setup? tks, -- Antonio J. S. Brandão

2 1

Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
by Sahria Hao 07 Feb '07

07 Feb '07

Hi Edson. I want to know that why my Cisco AS 5300 didn't send BYE for SER...? Maybe... I doubt that maybe my 5300 have only dial-peer 6000 voice "POTS" configure for outgoing PSTN call. In case of PSTN incoming call have no problem about sending BYE for SER, Because it is apply dial-peer voice 5000 "VOIP" confiure as follows: [SER] <- [5300 (VOIP dial-peer)] <- [PSTN] So I'll try to re-configure my 5300 dial-peer, or please give me a hint If anyone have some way to solve this problem. Thanks, Sahria 2007/2/6, Edson <4lists(a)gmail.com>: > > I have this same behaviour, but never give it great importance, since we > didn't bill incomming calls… > > But it would be great to know if it's because of a misconfiguration or a > bug… but we notice that many ports become unavaliable (blocked) over time. > To release we programmed a reboot every day on 3AM…J > > Even with 'ngrep' the BYE, when PSTN side disconnects, didn't show up… > > Edson. > > ------------------------------ > > *From:* serusers-bounces(a)lists.iptel.org [mailto: > serusers-bounces(a)lists.iptel.org] *On Behalf Of *Sahria Hao > *Sent:* segunda-feira, 5 de fevereiro de 2007 08:36 > *To:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's > bug? > > Hi Greger, > > And I'm very sorry for my poor exposition. > > >>Do you get an error on the 5300? > > No, my 5300 works well and there's no error. > > >> Is it sent, but never reaches SER? > > No, when I finished call by PSTN side, 5300 didn't send BYE for SER. > >>Does SER receive, but does not recognize it? > > SER didn't receive a message from 5300 entirely. > > I think that when I finished this call, 5300 must send a BYE message for > SER... but didn't send it. > 2007/2/5, Greger V. Teigre <greger(a)teigre.com>: > > 09. [Cisco] can't send BYE for SER *****why??***** > > What does that mean?! Do you get an error on the 5300? Is it sent, but > never reaches SER? > Does SER receive, but does not recognize it? > g-) > > Sho Aihara wrote: > > Hi all. > > I have a problem for the following scenario. > When I make a call for PSTN and on hook by PSTN side, > Cisco As can't send BYE for SER. > > 01. [UA via Asterisk] dialing "08022223333" -> [SER] > 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") > -> [Cisco] > 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from > "033308022223333" to "008022223333" > 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] > 05. [e.g. Mobile] Catch call > 06. [SER] log CDR start > 07. [Cisco] talking > 08. [e.g. Mobile] On hook and call disconnect > 09. [Cisco] can't send BYE for SER *****why??***** > 10. [UA via Asterisk] On hook > 11. [UA via Asterisk] Send BYE for SER > 12. [SER] log CDR End [Cisco] Call finished > > But another scenario, if make a call from PSTN to Asterisk and > on hook by PSTN side, Cisco As send BYE to SER. > > 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" > 02. [Cisco] receive "77771111" call number > 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: > my.ser.ip.address -> [SER] > 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] > 05. [UA via Asterisk] Catch call > 06. [SER] log CDR start > 07. [UA via Asterisk] talking > 08. [e.g. Mobile] On hook and call disconnect > 09. [Cisco] Send BYE to SER > 10. [SER] log CDR End [Cisco] Call finished > 11. [UA via Asterisk] receive BYE from SER > > And sorry for my diffucult example. > > Why Cisco AS 5300 can't send BYE to SER > When PSTN call is disconnected by PSTN side? > > My ser.cfg as follows: > > # > -------------------------------------------------------------------------- > # global configuration parameters > # > -------------------------------------------------------------------------- > fork=no > log_stderror=yes > check_via=no > dns=no > rev_dns=no > listen=my.ser.ip.address > port=5060 > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser:heslo@localhost/ser" > > # > -------------------------------------------------------------------------- > # module loading > # > -------------------------------------------------------------------------- > loadmodule "/usr/local/lib/ser/modules/mysql.so" > loadmodule "/usr/local/lib/ser/modules/sl.so" > loadmodule "/usr/local/lib/ser/modules/tm.so" > loadmodule "/usr/local/lib/ser/modules/rr.so" > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > loadmodule "/usr/local/lib/ser/modules/usrloc.so" > loadmodule "/usr/local/lib/ser/modules/registrar.so" > loadmodule "/usr/local/lib/ser/modules/textops.so" > loadmodule "/usr/local/lib/ser/modules/auth.so" > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > loadmodule "/usr/local/lib/ser/modules/avpops.so" > loadmodule "/usr/local/lib/ser/modules/permissions.so" > loadmodule "/usr/local/lib/ser/modules/acc.so" > loadmodule "/usr/local/lib/ser/modules/exec.so" > > # > -------------------------------------------------------------------------- > # setting module-specific parameters > # > -------------------------------------------------------------------------- > modparam("usrloc", "db_mode", 2) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("rr", "enable_full_lr", 1) > modparam("usrloc", "db_url", " mysql://ser:heslo@localhost/ser") > modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("permissions", "db_url", "mysql://ser:heslo@localhost /ser") > modparam("tm", "fr_inv_timer", 27) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("acc", "db_flag", 2) > modparam("acc", "db_missed_flag", 3) > > # > -------------------------------------------------------------------------- > # route pattern > # > -------------------------------------------------------------------------- > route { > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > break; > }; > > record_route(); > > if (loose_route()) { > if (method=="ACK") { > acc_db_request("01:CallStart\n", "acc"); > }; > if (method=="BYE" || method=="CANCEL") { > acc_db_request("02:CallEnd\n", "acc"); > }; > t_relay(); > break; > }; > > if (uri==myself) { > if (method=="REGISTER") { > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > break; > }; > save("location"); > break; > }; > > if (search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)%28my%5C.cisco%5C.ip%5C.address>)")) > { > #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 > rewritehost("my.asterisk.ip.address "); > }; > > lookup("aliases"); > > if (!lookup("location")) { > if (method=="INVITE" && !search("^(f|From): > .*(a)(my\.cisco\.ip\.address <.*(a)%28my%5C.cisco%5C.ip%5C.address>)")) { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > break; > }; > if (uri=~"^sip:0[0-9]{10}@") { > # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 > prefix("0333"); > rewritehostport("my.cisco.ip.address:5060"); > avp_write("i:45", "inv_timeout"); > } else { > sl_send_reply("404", "Not Found"); > break; > }; > consume_credentials(); > }; > }; > > }; > > if (!t_relay()) { > sl_reply_error(); > }; > > } > > And my Cisco AS 5300 config as follows: > > voice call send-alert > voice rtp send-recv > > voice service pots > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco > > voice service voip > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco > sip > min-se 60 > > translation-rule 50 > Rule 0 0333 0 > Rule 1 ^7777 037777 > > voice class codec 2 > codec preference 1 g711ulaw > codec preference 2 g711alaw > > dial-peer voice 5000 voip > tone ringback alert-no-PI > description ser-asterisk-cisco-test > huntstop > destination-pattern 77771111$ > translate-outgoing called 50 > voice-class codec 2 > session protocol sipv2 > session target ipv4:my.ser.ip.address > dtmf-relay rtp-nte > max-conn 1 > > dial-peer voice 6000 pots > application session > max-conn 2 > destination-pattern 0333T > progress_ind alert enable 8 > translate-outgoing called 50 > port 0:D > > Thanks, > Sahria > > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > -- > ---------- > shosuke > msn : anseie(a)hotmail.co.jp > email : sahria.hao(a)gmail.com >

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[Users] distinguish between fr_timer and fr_inv_timer timeout
by Klaus Darilion 07 Feb '07

07 Feb '07

Hi! If a failure route is executed - how can I distinguish between fr_timer timeout (a broken gateway) and fr_inv_timer timeout (nobdy took up the phone)? thanks klaus

3 4

[Users] 180 Ringing / 183 Session Progress
by Sam Lee 07 Feb '07

07 Feb '07

Hey Guys, Is it mandatory for SIP servers to generate to UA 180 Ringing to inform them that the called party is now ringing ? The reason i'm asking is because I am seeing strange results using different UAs. Using SIP clients to call out , instead of seeing 180 Ringing , i see 183 session progress even before call was picked up. On some sip client , ring back can be heard. On some others , ring back cannot be heard. Is there any reason to suspect that these UAs needs to receive 180 Ringing before generating any ringback ? Regards, Sam

4 6

[Users] OpenSER v1.0.1 to v1.1.1
by Max Gregorian 06 Feb '07

06 Feb '07

Hi, with regards to the release of v.1.1.1 I have a few questions. Firstly, what is to become of v1.0.1? Are you recommending all users with 1.0.1 also upgrade directly to v.1.1.1? Secondly, does 1.0.1 suffer from most (if not all) the same bugs as 1.1.0? I am particularly interested in any documented issues with handling response ACKs for 4xx class errors as I am currently experiencing a few issues with these, and I noticed in the changelog for 1.1.1 that a bug with ACK matching for non-200 class responses has recently been fixed. I ask this I noticed you are gradually phasing out all documentation to do with 1.0.1 on the openser website. As such, is 1.0.1 to be deprecated?

2 3

[Serusers] Has anyone got Jabber module to work w/SER
by SER LIST 06 Feb '07

06 Feb '07

Subject says it all. I am trying to make this work. I have installed JABBERD2 with a PyMSNT gateway. I can use a jabber cliient (exodus) and connect to Jabberd server and communicate w/ other Jabber users and MSN users. I now want to forward IM messages from Eyebeam UA to Jabber users and MSN users. I followed all instructions. When the message gets forwarded to SER and it executes the "xjab" function, I get an error saying "Invalid Content-Type". Any pointers will be highly appreciated. _________________________________________________________________ Buy what you want when you want it on Sympatico / MSN Shopping http://shopping.sympatico.msn.ca/content/shp/?ctId=2,ptnrid=176,ptnrdata=08…

1 0

[Users] avp_db_load question
by Klaus Darilion 06 Feb '07

06 Feb '07

Hi! What are the return values of avp_db_load? E.g. is it FALSE on problems accessing the DB or FALSE if it did not found any AVP to load? I couldn't find this documented somewhere. regards klaus -- Klaus Darilion nic.at

2 1

[Serusers] Anyone using CDRTool ?
by Lokesh Kumar 06 Feb '07

06 Feb '07

Hello All, Is there anyone using CDRTool in conjugation with ser, asterisk, radius setup. Can anyone give me a hand to setup it properly? Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.27/671 - Release Date: 2/5/2007

1 0

[Serusers] Request for Daniel-Constantin Mierla
by Giovanni Chirico 06 Feb '07

06 Feb '07

Hi Daniel, Im trying to use the dispatcher module. Im using the call forwarding ser.cfg file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack. I have inserted the function ds_select_dst("1", "0"); In the main route() and the function forward(uri:host, uri:port); in the PSTN Handler, route(5). How I can made the failure route, so if a gateway is down another gateway is load? In attachment there are my ser.cfg and dispatcher.list files. Please help me :-( Bye, Giò

2 3

[Serusers] automated response
by Myron Groening 06 Feb '07

06 Feb '07

Vacation Alert !! I will be out of the office, and will not be checking my message till FEB. 26 2007 Thanks Myron Groening

1 0

[Serusers] new version of radiusclient-ng
by Jan Janak 06 Feb '07

06 Feb '07

Hello, I just released a new version of radiusclient-ng library. The new version, 0.5.4, fixes problems on 64-bit architectures. You can download the new release from http://ftp.iptel.org/pub/radiusclient-ng/radiusclient-ng-0.5.4.tar.gz Credits go to Peter Nixon. Jan.

2 4

[Users] ERROR: module version mismatch for,/usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls,(i386/freebsd); module: openser 1.1.1-notls (i386/freebsd)
by Hansi Lao 06 Feb '07

06 Feb '07

Hello All, I have installed openser-1.1.1 with tls on my box. However, when I try to run openser. It returns me this error: ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd). Would someone be kind enough to help me how to compile a mysql.so which supports tls? Thank you.

2 4

[Users] migrating openser v1.1.x to v1.2.x
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, since we got in testing phase for 1.2.0 release, I think is better to start early a document with migration guidelines. Issues encountered during upgrade should be gathered together in the next page, to share and help each others: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x It is just the skeleton and few stuff there. Hopefully the content will grow quickly. Everybody can contribute. Cheers, Daniel

1 0

[Users] How can i make my OpenSER server with presence module work properly with the db_text database?
by Kc Ayuba 05 Feb '07

05 Feb '07

Hi, I'm trying to use db_text database with the presence module of OpenSER(1.2.x ). User agent is X-Lite(3.0). When I try to send a REGISTER message to OpenSER, OpenSER still works. But if i sent a SUBSCRIBE or a PUBLISH message, then openser crashes. If i use the MySQL database, then everything works just fine. But i prefer to use the db_text database. Here is the core file that was generated when OpenSER crashed. Core generated: Program terminated with signal 11, Segmentation fault. #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) bt #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 #1 0x080a3f6c in timer_ticker () at timer.c:141 #2 0x0806b51f in main_loop () at main.c:724 #3 0x0806d43b in main (argc=3, argv=0xaff0a2e4) at main.c:1438 (gdb) frame 0 #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) list 605 606 subs.from_domain.s = row_vals[from_domain_col].val.str_val.s; 607 subs.from_domain.len = strlen(row_vals[from_domain_col].val.str_val.s); 608 609 subs.event_id.s = row_vals[event_id_col].val.str_val.s; 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); 611 612 subs.to_tag.s = row_vals[to_tag_col].val.str_val.s; 613 subs.to_tag.len = strlen(row_vals[to_tag_col].val.str_val.s); 614 (gdb) 615 subs.from_tag.s = row_vals[from_tag_col].val.str_val.s; 616 subs.from_tag.len = strlen(row_vals[from_tag_col].val.str_val.s); 617 618 subs.callid.s = row_vals[callid_col].val.str_val.s; 619 subs.callid.len = strlen(row_vals[callid_col].val.str_val.s); 620 621 subs.contact.s = row_vals[contact_col].val.str_val.s; 622 subs.contact.len = strlen(row_vals[contact_col].val.str_val.s); 623 624 subs.cseq = row_vals[cseq_col].val.int_val; I really hope that someone can help me with this problem. Thanks in advance. Kind regards, Kc

4 9

RE: [Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
by Jon Webster 05 Feb '07

05 Feb '07

The significant difference of debug from a good call and openser is next. Below that are the full debug logs of both the openser and goodproxy call with an ngrep of openser's call in between. significant differences ======================== Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> VERSES --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: asterisk debug of openser call =========== --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 180 Ringing FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- Called openser/8886000 -- SIP/openser-082966e8 is ringing pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 200 OK FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM> CONTENT-LENGTH: 197 CONTENT-TYPE: application/sdp SERVER: RTCC/2.0.6017.0 v=0 o=- 0 0 IN IP4 exchangeUM s=Microsoft Exchange Speech Engine c=IN IP4 exchangeUM t=0 0 m=audio 30660 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> <-- SIP read from openser:5060: BYE sip:3149@asterisk SIP/2.0 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f TO: <sip:3149@asterisk>;tag=as4aa53078 CSEQ: 1 BYE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 69 Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825 VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 CONTENT-LENGTH: 0 USER-AGENT: RTCC/2.0.6017.0 P-hint: outbound --- (13 headers 0 lines) --- Sending to openser : 5060 (non-NAT) Transmitting (no NAT) to openser:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f To: <sip:3149@asterisk>;tag=as4aa53078 Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:3149@asterisk> Content-Length: 0 ngrep of above call ====================== # U +0.000616 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.067855 openser:5060 -> asterisk:5060 SIP/2.0 200 OK. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTACT: <sip:swordfish:5065;transport=Tcp;maddr=exchangeUM>. CONTENT-LENGTH: 197. CONTENT-TYPE: application/sdp. SERVER: RTCC/2.0.6017.0. . v=0. o=- 0 0 IN IP4 exchangeUM. s=Microsoft Exchange Speech Engine. c=IN IP4 exchangeUM. t=0 0. m=audio 30660 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U +32.353338 openser:5060 -> asterisk:5060 BYE sip:3149@asterisk SIP/2.0. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. TO: <sip:3149@asterisk>;tag=as4aa53078. CSEQ: 1 BYE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 69. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825. VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. CONTENT-LENGTH: 0. USER-AGENT: RTCC/2.0.6017.0. P-hint: outbound. . # U +0.000734 asterisk:5060 -> openser:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser. Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. To: <sip:3149@asterisk>;tag=as4aa53078. Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:3149@asterisk>. Content-Length: 0. . good call from working proxy ======== We're at asterisk port 17508 Video is at asterisk port 11130 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to goodproxy:5060: INVITE sip:6000@goodproxy SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Feb 2007 14:55:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 216 v=0 o=root 381 381 IN IP4 asterisk s=session c=IN IP4 asterisk t=0 0 m=audio 17508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called exchange12/6000 pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport;received=goodproxy From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: M-Networks USR/1.0 Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE Content-Length: 0 --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- SIP/exchange12-082b8e18 is ringing pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTACT: <sip:goodproxy:5060> CONTENT-LENGTH: 197 CONTENT-TYPE: application/sdp SERVER: RTCC/2.0.6017.0 v=0 o=- 0 0 IN IP4 exchangeUM s=Microsoft Exchange Speech Engine c=IN IP4 exchangeUM t=0 0 m=audio 47688 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: ACK sip:goodproxy:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK6c3f8b53;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/exchange12-082b8e18 answered SIP/jonlaptop-082b0fb8 -- Attempting native bridge of SIP/jonlaptop-082b0fb8 and SIP/exchange12-082b8e18 Scheduling destruction of call '109c82647d6340be2a5ab19e7eb18a14@asterisk' in 32000 ms set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set de5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK47d7f855;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (local, 7776000, 3) exited non-zero on 'SIP/jonlaptop-082b0fb8' pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK47d7f855;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;tag=f51839ab98;epid=BD-70-82-06-F9 CSEQ: 103 BYE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- Destroying call '109c82647d6340be2a5ab19e7eb18a14@asterisk' pbx*CLI>

1 0

[Users] CVS frozen - testing phase of 1.2.0
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, starting with today the development for 1.2.0 is closed. No new features should be added until the release. The next month will be dedicated to testing, integration of submitted patches and fixing reported bugs. We have a lot of new code to test, your help is very much appreciated. I want to list the major changes you should focus on while testing: - usage of pseudo-variables and transformations is script (http://openser.org/dokuwiki/doku.php/transformations:devel) - TM timers (millisecond accuracy) - IP blacklists - DNS failover - automatic error handling - flags system (script flags, message flags, branch flags) - usrloc enhancements (licking system refurbished, database content optimization via DB fetch support) - new management interface (mi_fifo, mi_xmlrpc) - SIP Servlet Java Application server interoperability (seas module) - PERL AGI (perl module) - SNMP support via snmpstats module - Presence support (modules presence, pua, pua_usrloc, pua_mi) - xmpp IM gateway (xmpp module) - sip session timer support (module sst) - accounting clean up - federation based peering via domainpolicy module If I forgot something, please contribute. In addition, great help would be with creating 1.2.0 variants of documentation and dokuwiki pages (where is the case), testing the tools (openserctl, DB creation scripts) and developing scripts to migrate database from 1.1.x to 1.2.0. To keep track of discovered issues, please register them on the tracker -- it this way you make sure it is not lost in mail threads: http://sourceforge.net/tracker/?group_id=139143&atid=743020 Have nice testing sessions! Your feedback is always helpful! Cheers, Daniel

2 3

[Serusers] DB disconnects with MySQL 5.0.27
by sip 05 Feb '07

05 Feb '07

Something I've never run into before when I was using MySQL 4.x, but now crops up on MySQL 5.0.27... I get these errors: Feb 5 07:30:40 qabalah ser[26765]: submit_query(): MySQL server has gone away Feb 5 07:30:40 qabalah ser[26765]: db_update: Error while submitting query Feb 5 07:30:40 qabalah ser[26765]: db_upd_ucontact(): Error while updating database Feb 5 07:30:40 qabalah ser[26765]: wb_timer(): Error while updating contact in db I assume there's some sort of timeout of the SER connection to mysql? Is there a standard way to rectify this? Is this something new in MySQL 5? N.

1 1

[Users] Socket Errors
by Martin Burns 05 Feb '07

05 Feb '07

We have over 4 proxies running openser-1.1.0-notls. Recently we started to see the following error in the logs of one of the proxies: Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: ERROR: udp_send: sendto(sock,0xb3c35898,594,0,0xb3c257f4,16): Invalid argument(22) Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: msg_send: ERROR: udp_send failed Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: ERROR:tm:t_forward_nonack: sending request failed Now the errors suggests a binding to localhost, however our openser.cfg has the following: listen=eth0:5060 After a brief search we came across a couple of over people having a similar problem. Once of the suggested fixes was to add a mhomed=1 line to the cfg. Well we tried this (even though we are not multi homed) and the error changed to: Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR: get_out_socket: connect failed: Invalid argument Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR: get_out_socket: no socket found Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR:tm:add_uac: can't fwd to af 2, proto 1 (no corresponding listening socket) Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR:tm:t_forward_nonack: failure to add branches Does anyone have any suggestions or ideas as to what is causing this error. Thanks mburns

2 7

[Users] dbtext script
by Ovidiu Sas 05 Feb '07

05 Feb '07

Hi, Since dbtext is part of the standard, my suggestion would be to ship/install the sc.dbtext script along with the openserctl script. We can install it as openser_dbtext_ctl and in openserctl we can make a reference to it if no DBENGINE is provided: see the following patch for openserct (openser-v1.1.1): diff -Naurp sip-server/scripts/openserctl openser/scripts/openserctl --- sip-server/scripts/openserctl 2006-09-15 08:41:42.000000000 -0400 +++ openser/scripts/openserctl 2007-01-17 09:39:09.000000000 -0500 @@ -126,6 +126,8 @@ do $f done echo + echo "Note: Use openser_dbtext_ctl for DBTEXT operations!" + echo } Regards, Ovidiu Sas

2 3

[Users] auth_radius.so - ERROR: load_module: could not open module
by Luca Corti 05 Feb '07

05 Feb '07

Hello, I've built openser 1.1.1 from source with radius support. I installed radiusclient-ng. When I execute openser with to loadmodule auth_radius.so in my configuration I get: # openser 0(20268) ERROR: load_module: could not open module </usr/local/lib/openser/modules/auth_radius.so loadmodule >: /usr/local/lib/openser/modules/auth_radius.so loadmodule : cannot open shared object file: No such file or directory 0(20268) parse error (60,13-14): failed to load module 0(20268) parse error (60,14-15): syntax error 0(20268) parse error (60,14-15): ERROR: bad config file (3 errors) But the module is there. # ls -l /usr/local/lib/openser/modules/auth_radius.so -rwxr-xr-x 1 root staff 61369 Feb 5 10:33 /usr/local/lib/openser/modules/auth_radius.so This is really strange. ciao Luca

2 3

Re: [Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

have you checked this on the x-lite machine? regards, bogdan raviprakash sunkara wrote: > it didn't see any Ack packet is sending to SIP server, > > > On 2/5/07, *Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > I would say the X-lite does not obey the RR header - please try to get > an trace from the X-lite machine to see where the ACK is sent. > > regards, > bogdan > > raviprakash sunkara wrote: > > > Hi BogDan, > > > > Yes, I see Record-Route header in the 200 ok....... > > Please can u check the ngrepped of the X-lite > > with Hard phones iit is fine ........... > > Below one is ngrep from X-lite users, > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

1 0

[Users] default send-socket
by Klaus Darilion 05 Feb '07

05 Feb '07

Hi! What is the default send socket? Is it the first listen=..... statement in openser.cfg? Or can I manually set the default sendsocket? regards klaus

2 1

[Serusers] Re: load mysql.so failed
by Su Kerry 05 Feb '07

05 Feb '07

2 1

[Users] mediaproxy MySQL server has gone away error
by James Holden 05 Feb '07

05 Feb '07

Hi all, I'm having a problem with the mediaproxy rtp proxy. I'm using to do accounting into a MySQL database. After a period of inactivity, it loses it's connection to the MySQL server and the update queries begin to fail. Openser stays connected, however. The error in the logs is: error: accounting failed. MySQL error: (2006, 'MySQL server has gone away') I'm running: mediaproxy 1.8.0 openser-1.1.0 (using the RHEL3 rpm from the openser website) openser-mysql-1.1.0 (again using the openser RHEL3 rpm) Python-2.4.4 MySQL-python-1.2.1_p2 My Python skills aren't up to much. So far, I've tried added the following line into /usr/local/mediaproxy/modules/mysql.py just before the query call, to try and re-establish the connection: def query(self, query): self._query = query for c in range(0, 3): try: self.conn.db.ping() <<<<<<---- added by me return self.cursor.execute(query) except MySQLdb.OperationalError, why: if why[0] not in (CR.SERVER_GONE_ERROR, CR.SERVER_LOST): raise raise ... but it hasn't helped. Has anyone seen this before? Best regards, James -- James Holden AQ Limited Senior Systems Architect 13-15 Hunslet Road james.holden(a)uk.aql.com Leeds, LS10 1JQ 07989 702968 0800 389 5815 ===================================================== http://aql.com/documents/news/about_aql.pdf "The service has been perfect." BSI Inspectorate ltd "really really really useful. easy to use and reduces manpower." Fortnums "without a shadow of a doubt...excellent" Ruella James Plc ----------------------------------------------------- (aq) Limited 13-15 Hunslet Road Leeds LS10 1JQ 194 Harrogate Road Leeds LS7 4NZ http://www.aql.com Sales: 08707 449227 Fax: 08707 449228 PGP public key http://aql.co.uk/A-Q.txt Subject to our standard terms, found at http://aql.com/site/termsofservice.php ----------------------------------------------------- ****************************** CONFIDENTIALITY This e-mail is intended only for the use of the addressees named above and may be confidential or legally privileged. If you are not an addressee you must not read it and must not use any information contained in nor copy it nor inform any person other than (aq) limited or the addressees of its existence or contents. If you have received this email and are not a named addressee, please delete it and notify the (aq) limited customer service department via admin(a)a-q.co.uk. Many thanks. =========================================================

2 3

[Serusers] Dispatcher Module Help
by Giovanni Chirico 05 Feb '07

05 Feb '07

Hi, I would like to use the dispatcher module. I have configured it on my ser.cfg file for the PSTN calls and it works fine. However when the select gateway is unavailable the module don't switch to another gateway. Can you help me to write the "failure_route" so I can use another gateway when the first is unavailable? 'm using the "call forwarding ser.cfg" file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack (in attachment I'm sending you my ser.cfg and dispatcher.list files) Thanks in advance for your help. Bye, Giovanni Chirico

1 0

Re: [Users] strip_tail() string length?
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Just a typo correction On 02/05/07 13:54, Brandon Armstead wrote: > Daniel-Constantin Mierla wrote: >> Stripping is now possible with the new introduced 'transformations': >> http://openser.org/dokuwiki/doku.php/transformations:devel >> >> So if you want to strip last two characters of username in R-URI >> >> $var(x) = $(rU{s.len})-2; >> $rU = $(ru{s.substr,0,$var(x)}); ^^^ this is: $rU = $(rU{s.substr,0,$var(x)}); Cheers, Daniel >> >> Cheers, >> Daniel >> >> >> On 01/30/07 09:08, Juha Heinanen wrote: >>> Brandon Armstead writes: >>> >>> > i.e. strip_tail(strlen($ru) - 2); ... or something like such. >>> >>> sorry, but my suggestion >>> >>> substr($ru, -2, 2) >>> >>> does not work, since it returns the last two chars of $ru not the >>> remaining chars. strlen is indeed needed for the above. using php >>> substr and strlen >>> >>> strip_tail(strlen($ru) - 2) >>> >>> becomes >>> >>> substr($ru, 0, strlen($ru) - 2) >>> >>> -- juha >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > Hello Daniel, > > Glad to see this implemented, I think it will be a definite asset > to the future and improvements of OpenSER. Keep up the good work! > > Regards, > Brandon >

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Re: [Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

Ravi, so you see any Record-Route hdrs in the 200 OK reply? regards, bogdan raviprakash sunkara wrote: > Hi BogDan, > Good AfterNoon, and Thanks For replying to me, > > I used the Following code for Record_route(); > > route { > .... > ..... > if(method=="INVITE" && nat_uac_test("3") ) > { > record_route_preset("sip: > 192.168.2.3:5060 <http://192.168.2.3:5060>;nat=yes"); > }else if ( method ! = " REGISTER ") > { > record_route(); > } > } > where 192.168.2.3:5060 <http://192.168.2.3:5060> is the private IP of > the SIP server behind the NAT. > > > On 2/5/07, * Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Ravi, > > do you do record_route() for the INVITE? if not, the sequential > requests > of the dialog will bypass your server. > > regards, > bogdan > > raviprakash sunkara wrote: > > > Hello Users, > > > > When I tested the OpenSER server with CISCO ATA 186/188 IP phones > > SIP EyeP media (Softphnoes ) is working fine with RTP , when Both > > Users and SIP Server is in Behind the NAT, > > > > But Problem with Softphones ( X-lite ) . > > > > When I ngrep the Calls of X-lite users i didn't see any Ack > Packet to > > OpenSER After the Invite Packets received . > > > > Please Help me in this Part > > > > > > -- > > Thanks and Regards > > Ravi Prakash Sunkara > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> > <mailto:ravi.sunkara@hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > M:+91 9985077535 > > O:+91 40 23114549 > > F:+91 40 40208727 > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> <mailto: > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > www.hyperion-tech.com <http://www.hyperion-tech.com> > <http://www.hyperion-tech.com> > > > >------------------------------------------------------------------------ > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

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Re: [Users] 200 ok not cought in ONREPLY
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

Dan, that is rather strange if you are confident you are setting correctly a reply route (not default one) for the INVITE transaction. add the following debug message in modules/tm/t_reply.c, after line 1312: DBG("---------onreply route is %u\n",t->on_reply); recompile ans see the output on runtime. regards, bogdan Dan-Cristian Bogos wrote: > Salut Bogdan, > > theoretically I have. Saw now that with default onreply I can catch it > (log). It is really strange since I am relying both BYE and INVITES in > the same t_relay block with t_on_reply("1"); > t_on_failure("1"); but the 200 OK comes to default. > > Cheers, > Dan > > On 2/5/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >> Salut Dan, >> >> are you sure you have an onreply route sent for this transactions? from >> the output I see no problems. >> >> regards, >> bogdan >> >> Dan-Cristian Bogos wrote: >> >> > Hi Guys, >> > >> > By receiving incoming traffic from some other sip server (configured >> > for different domain than mines) I will not be able to catch in the >> > routing, onreply section, the 200 OK for the INVITE , OK which my >> > client, behind my server, is sending to remote side, therefore not >> > beeing able to modify sdp to use mediaproxy. Bellow is a level8 of >> > debug of the OK. >> > Is there any other way of catching this message in the routing? Any >> > help will be appreciated. >> > >> > Thxs in advance, >> > Dan >> > >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply >> > (status): >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: >> > <SIP/2.0> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: >> <200> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: >> <OK> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the first via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After >> > parse_msg... >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module nathelper, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the second via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 234, <received> = <87.139.12.167>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 235, <rport> = <5096>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bK521554023>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_param: tag=8b2d7e45 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:parse_to:end of header reached, state=29 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to >> > body [<sip:dan@de.babble.net>] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > get_hdr_field: cseq <CSeq>: <20> <INVITE> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module tm, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=0 T start=0xffffffff >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=22 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=8 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: hash 39598 label 705981201 branch 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:REF_UNSAFE: after is 1 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: reply matched (T=0x4081f430)! >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=1 T end=0x4081f430 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 >> > is_invite=1) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: branch=0, save=0, relay=0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: >> > 911, new size: 850 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= >> > SIP/2.0 200 OK^M Via: SIP/2.0/UDP >> > 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP >> > >> 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M >> >> > >> > Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: >> > <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: >> > <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: >> > <sip:dan@de.babble.net>;tag=8b2d7e45^M From: >> > <sip:dan@babble.net>;tag=2240571840^M Call-ID: >> > 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, >> > CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M >> > Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp >> > 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M >> > s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio >> > 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 >> > telephone-event/8000^M a=sendrecv^M >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_to_tail_of_timer[2]: 0x4081f478 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., >> > shmem=0x40821528: SIP/2.0 2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > cleanup_uac_timers: RETR/FR timers reset >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:UNREF_UNSAFE: after is 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:destroy_avp_list: destroying list (nil) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: >> > cleaning up >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >> >

1 0

[Users] Hijack 'Move Temporarily'
by Sam Lee 05 Feb '07

05 Feb '07

Hey Guys, Normally if the SIP-UA is call-forwarded to a different number , it will send a 'Move Temporarily' to the proxy , which in turn relay this message to the originating gateway. The problem is , if they call forward to a number which the originating gateway doesn't know what to do with it , it will just kill the call. Can i hijack 'Move Temporarily' directly in the proxy and sending this call to a different gateway rather than relaying it to the originating gateway ? Please let me know if you can't understand me. Thanks! Regards, Sam

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[Users] database persistent connection
by tele 05 Feb '07

05 Feb '07

Hi, I've an annoying problem with database persistent connection. I'm using mysql database for location and avp stuff. During the day all it's but after a day i retest some call and seams that the proxy lost connection with the database, and avp and location stuff didn't works anymore. I'm using this version of openser: # openser -V version: openser 1.2.0-dev8-notls (i386/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, select, kqueue. @(#) $Id: main.c,v 1.23 2006/10/02 15:51:45 bogdan_iancu Exp $ main.c compiled on 12:38:52 Nov 27 2006 with gcc 3.4.4 i've also configure a small ping_interval for the mysql module modparam("mysql", "ping_interval", 60) modparam("mysql", "auto_reconnect", 1) and the usrloc: modparam("usrloc", "db_mode", 3) modparam("usrloc", "matching_mode", 1) I've also check that the NTP is correct on the servers. I don't know why openser lost connection with the database. any hints? thanks :tele

2 2

[Users] Help. in OpenSER in Post-paid and pre-paid
by raviprakash sunkara 05 Feb '07

05 Feb '07

Hello Users, I'm Using The Following software of My Sip productions, 1) OpenSER is a Sip Server,proxy, in front-end 2) Asterisk is PBX server, in Back-end 3) Radius for AAA to OpenSER, 4) Asterisk-java for IVR for Billing Customer care if I did any mistakes in these mail , forgive me . My Question is 1) If Caller calls to Callee , Before the call established to callee. The Caller has to know his Balance Account. if the Balance is there, then only the call has to establish, 2) I want the OpenSER total in RealTime application Server like Java application . Call Authorization ( Question no 1 ). If Asterisk is Front-end , ---> problem is Solved, How can a b2bua implement with OpenSER, please when -- 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

2 2

[Users] strip_tail() string length?
by Brandon Armstead 05 Feb '07

05 Feb '07

Hello everyone, I'm wondering what the easiest method would be to grab how long the RURI is and strip based on that information... or would I have to actually use regex, or is there an easier STRLEN() or LENGTH() operation? Any ideas, or is this something that could be implemented? Thanks. i.e. strip_tail(strlen($ru) - 2); ... or something like such. -- Brandon Armstead

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[Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
by Sho Aihara 05 Feb '07

05 Feb '07

Hi all. I have a problem for the following scenario. When I make a call for PSTN and on hook by PSTN side, Cisco As can't send BYE for SER. 01. [UA via Asterisk] dialing "08022223333" -> [SER] 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco] 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333" 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] 05. [e.g. Mobile] Catch call 06. [SER] log CDR start 07. [Cisco] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] can't send BYE for SER *****why??***** 10. [UA via Asterisk] On hook 11. [UA via Asterisk] Send BYE for SER 12. [SER] log CDR End [Cisco] Call finished But another scenario, if make a call from PSTN to Asterisk and on hook by PSTN side, Cisco As send BYE to SER. 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" 02. [Cisco] receive "77771111" call number 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER] 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] 05. [UA via Asterisk] Catch call 06. [SER] log CDR start 07. [UA via Asterisk] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] Send BYE to SER 10. [SER] log CDR End [Cisco] Call finished 11. [UA via Asterisk] receive BYE from SER And sorry for my diffucult example. Why Cisco AS 5300 can't send BYE to SER When PSTN call is disconnected by PSTN side? My ser.cfg as follows: # -------------------------------------------------------------------------- # global configuration parameters # -------------------------------------------------------------------------- fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=my.ser.ip.address port=5060 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # -------------------------------------------------------------------------- # module loading # -------------------------------------------------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # -------------------------------------------------------------------------- # setting module-specific parameters # -------------------------------------------------------------------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -------------------------------------------------------------------------- # route pattern # -------------------------------------------------------------------------- route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { if (method=="ACK") { acc_db_request("01:CallStart\n", "acc"); }; if (method=="BYE" || method=="CANCEL") { acc_db_request("02:CallEnd\n", "acc"); }; t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (search("^(f|From): .*(a)(my\.cisco\.ip\.address)")) { #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 rewritehost("my.asterisk.ip.address"); }; lookup("aliases"); if (!lookup("location")) { if (method=="INVITE" && !search("^(f|From): .*(a)(my\.cisco\.ip\.address)")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (uri=~"^sip:0[0-9]{10}@") { # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 prefix("0333"); rewritehostport("my.cisco.ip.address:5060"); avp_write("i:45", "inv_timeout"); } else { sl_send_reply("404", "Not Found"); break; }; consume_credentials(); }; }; }; if (!t_relay()) { sl_reply_error(); }; } And my Cisco AS 5300 config as follows: voice call send-alert voice rtp send-recv voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip min-se 60 translation-rule 50 Rule 0 0333 0 Rule 1 ^7777 037777 voice class codec 2 codec preference 1 g711ulaw codec preference 2 g711alaw dial-peer voice 5000 voip tone ringback alert-no-PI description ser-asterisk-cisco-test huntstop destination-pattern 77771111$ translate-outgoing called 50 voice-class codec 2 session protocol sipv2 session target ipv4:my.ser.ip.address dtmf-relay rtp-nte max-conn 1 dial-peer voice 6000 pots application session max-conn 2 destination-pattern 0333T progress_ind alert enable 8 translate-outgoing called 50 port 0:D Thanks, Sahria

3 2

Re: [Users] 200 ok not cought in ONREPLY
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, On 02/05/07 12:30, Dan-Cristian Bogos wrote: > Dear Daniel, > > many thanks for the reply. I can perfectly catch the OK now. ok > > The only thing which I must find out is why for a normal INVITE the OK > was not coming in the same onreply block as the BYE, since both INVITE > and BYE are relied in the same way out and processed theoretically in > the same route block. BYE is usually following the loose routing -- but I do not know if it is the case of your comfig. Cheers, Daniel > > Anyway, thank you again, it was really great help. > > Cheers, > Dan > > > On 2/5/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> to get a reply in an onreply_route other than the default on, you have >> to arm it via t_on_reply() when you relay the request. >> ... >> t_on_reply("1"); >> ... >> >> t_on_reply[1] { >> } >> >> Otherwise, all replies are processed in the default onreply_route (the >> one which has no index) >> >> onreply_route { >> .... >> } >> >> Cheers, >> Daniel >> >> On 02/01/07 15:48, Dan-Cristian Bogos wrote: >> > Hi Guys, >> > >> > By receiving incoming traffic from some other sip server (configured >> > for different domain than mines) I will not be able to catch in the >> > routing, onreply section, the 200 OK for the INVITE , OK which my >> > client, behind my server, is sending to remote side, therefore not >> > beeing able to modify sdp to use mediaproxy. Bellow is a level8 of >> > debug of the OK. >> > Is there any other way of catching this message in the routing? Any >> > help will be appreciated. >> > >> > Thxs in advance, >> > Dan >> > >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply >> > (status): >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: >> > <SIP/2.0> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: >> <200> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: >> <OK> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the first via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After >> > parse_msg... >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module nathelper, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the second via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 234, <received> = <87.139.12.167>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 235, <rport> = <5096>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bK521554023>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_param: tag=8b2d7e45 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:parse_to:end of header reached, state=29 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to >> > body [<sip:dan@de.babble.net>] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > get_hdr_field: cseq <CSeq>: <20> <INVITE> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module tm, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=0 T start=0xffffffff >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=22 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=8 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: hash 39598 label 705981201 branch 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:REF_UNSAFE: after is 1 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: reply matched (T=0x4081f430)! >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=1 T end=0x4081f430 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 >> > is_invite=1) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: branch=0, save=0, relay=0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: >> > 911, new size: 850 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= >> > SIP/2.0 200 OK^M Via: SIP/2.0/UDP >> > 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP >> > >> 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M >> >> > >> > Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: >> > <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: >> > <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: >> > <sip:dan@de.babble.net>;tag=8b2d7e45^M From: >> > <sip:dan@babble.net>;tag=2240571840^M Call-ID: >> > 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, >> > CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M >> > Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp >> > 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M >> > s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio >> > 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 >> > telephone-event/8000^M a=sendrecv^M >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_to_tail_of_timer[2]: 0x4081f478 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., >> > shmem=0x40821528: SIP/2.0 2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > cleanup_uac_timers: RETR/FR timers reset >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:UNREF_UNSAFE: after is 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:destroy_avp_list: destroying list (nil) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: >> > cleaning up >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >

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