- sr-users - kamailio.org

[OpenSER-Users] X-ten failed to make connection
by Michael Chan 29 Feb '08

29 Feb '08

Derek Please see the attached mailing address and send them the infor for help. MC

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[OpenSER-Users] Why $rP is not writeable?
by Iñaki Baz Castillo 29 Feb '08

29 Feb '08

Hi, sometimes I need to forward a message to a SIP UAS that just speaks UDP (for example a PSTN gateway), so if I recevie an INVITE from a TCP client I must use: t_relay("udp:gateway:5060") since the following would fail: $rd = "gateway"; t_relay(); The problem is that this second approach is more flexible and it can be used in block routes, being them independent fo the final destination (set prevously with asignements like "$rd = gateway"). A solution for this could be allow $rP: $rP - reference to transport protocol of R-URI so I could do: $rP = "UDP"; $rd = "gateway"; t_relay(); But unfortunatelly $rP is no writeable: http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#assignment Another solution could be if "force_send_socket" function would allow setting **just** the protocol type, but it doesn't allow it. So there is no solution excepting making dirty the script and setting all the destination (protocol:ip:port) using "t_relay(protocol:ip:port)". Why $rP is no writeable? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] mediaproxy server NEW FEATURE!!!
by Gonzalo J. Sambucaro 29 Feb '08

29 Feb '08

Raju, Thanks for testing it, now I am working in making it more secure. Regards, Gonzalo. > Gonzalo, > > Thanks!! It works very well esp. we have few clients > behind DSL which is unstable and the connectivity is > remarkably improved. > > Good contribution to mediaproxy. > > regards, > > Raju > --- "Gonzalo J. Sambucaro" > <gonzalo.sambucaro(a)mslc.com.ar> wrote: > >> Raju, >> please replase the file rtphandler.py. >> >> Regards, >> Gonzalo. >> >> > Hi Gonzalo, >> > >> > Can you please let me know how I can try out this >> > addition. I am currently using Mediaproxy 1.9.1 >> along >> > with openser-tls-1.1.1. >> > >> > >> > Please do let me know where I can get this patch. >> > >> > regards, >> > >> > Raju >> > --- "Gonzalo J. Sambucaro" >> > <gonzalo.sambucaro(a)mslc.com.ar> wrote: >> > >> >> Andreas, >> >> our local ISP´s are disconnecting DSL >> lines >> >> more ofthen but it >> >> renews the IP almost instantaneously. >> >> >> >> Regards, >> >> Gonzalo. >> >> >> >> > Hello, >> >> > nice feature, i know the problem, because our >> >> local ISP´s are >> >> > disconnecting DSL lines every 8 hours. >> >> > But the DSL line needs nearly 8 seconds, to be >> up >> >> and running again(PPP >> >> > dialup), if i can´t hear my >> >> > partner for a few seconds i will disconnect the >> >> line and redial, my >> >> > opinion. >> >> > >> >> > Same in mobile environments, when the handover >> >> doesn´t work, saying >> >> > "hello? - hello?" -> hangup -> >> >> > redial. >> >> > >> >> > But i´m willing to try, leave me a note, where >> i >> >> can download your >> >> > version. >> >> > >> >> > regards, >> >> > Andreas M. >> >> > >> >> > >> >> > >> >> > Gonzalo J. Sambucaro schrieb: >> >> >> Folks, >> >> >> I developed a new feature in the Media >> >> Proxy project of >> >> >> AG-Proyect. >> >> >> With this new feature the Media Proxy Server >> is >> >> now able to manage the >> >> >> problem of the NAT IP change in an established >> >> call. It is >> >> >> currently in production and working without >> any >> >> problem. >> >> >> >> >> >> Somebody wants to test it? >> >> >> >> >> >> I would like to know what to do to make this >> >> feature available to >> >> >> everybody. Does anybody know? >> >> >> >> >> >> Regards, >> >> >> >> >> >> ------------------------------ >> >> >> >> >> >> Details: >> >> >> ------- >> >> >> Supose that there is an established rtp >> >> connection between two >> >> >> endpoints and the media proxy is in the middle >> >> doing the relay of >> >> >> the rtp streams, let´s say >> >> >> >> >> >> (MP = mediaproxy) >> >> >> >> >> >> EP A <--------->[NAT with IP1]<-----------> MP >> >> <--------------------> EP >> >> >> B >> >> >> >> >> >> EP A sends rtp to MP_IP:MP_PORT passing >> through >> >> the NAT box. >> >> >> EP B sends rtp to MP_IP:MP_PORT without >> passing >> >> through a NAT box. >> >> >> >> >> >> The MP know that the caller = >> NAT_IP1:NAT_PORT1, >> >> and the called = >> >> >> EP_B_IP:EP_B_PORT >> >> >> >> >> >> Now, supose that the NAT box change their >> PUBLIC >> >> IP from IP1 to IP2, so >> >> >> this escenary >> >> >> >> >> >> EP A <------->[NAT with IP1]<---------> MP >> >> <-------------> EP B >> >> >> >> >> >> will change to this >> >> >> >> >> >> EP A <------->[NAT with IP2]<---------> MP >> >> <--------------> EP B >> >> >> >> >> >> so the MP should detect that change of IPs and >> >> continue relaying the rtp >> >> >> streams but now to IP2:PORT2 instead of >> >> IP1:PORT1. >> >> >> >> >> >> Well, that was the situation y have >> experienced. >> >> >> >> >> >> To fix this, I developed this solution, >> changing >> >> rtphandler.py file: >> >> >> >> >> >> 1) When the first rtp packet of a source >> arrives, >> >> save the SSRC field in >> >> >> the MP. >> >> >> - Save the SSRC of the caller. >> >> >> - Save the SSRC of the called. >> >> >> >> >> >> 2) If arrives a rtp packet with unknown source >> IP >> >> but with the same SSRC >> >> >> field of some of the two streams, updates the >> >> binding (with the new IP >> >> >> detected) between the caller and the MP or >> >> between the called and the MP >> >> >> according to the field SSRC previously saved. >> >> >> >> >> >> Note: SSRC (RFC 3550 RTP), (from the rfc: >> "The >> >> SSRC identifier carried >> >> >> in >> >> >> the RTP header and in various fields of RTCP >> >> packets is a random 32-bit >> >> >> number that is required to be globally unique >> >> within an RTP session ") >> >> >> >> >> >> -------------------------------- >> >> >> >> >> > >> >> > -- >> >> > g, >> >> > Andreas M. >> >> > >> >> > _______________________________________________ >> >> > Users mailing list >> >> > Users(a)lists.openser.org >> >> > >> >> >> > >> > http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > >> >> >> >> >> >> -- >> >> Gonzalo J. Sambucaro >> >> Ingeniería de Software >> >> Tel: +54-341-4230504 >> >> MSLC >> >> gonzalo.sambucaro(a)mslc.com.ar >> >> www.mslc.com.ar >> >> Ocampo y Esmeralda - Vivero de Empresas de Base >> >> Tecnológica >> >> Ciudad Universitaria Rosario UNR, CCT CONICET >> >> Rosario - Santa Fé - Argentina >> >> >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users(a)lists.openser.org >> >> >> > >> > http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> >> > >> > >> >> >> -- >> Gonzalo J. Sambucaro >> > === message truncated ===> >> ''' >> Handle media stream relaying and provide control >> ports to receive and >> process requests from SER (SIP Express Router) or >> the proxydispatcher. >> >> Copyright 2003-2005 Dan Pascu >> ''' >> >> import asyncore >> import socket >> import sys, os, grp, errno, traceback, atexit >> import time >> import struct >> import re >> import weakref >> >> from utilities import * >> from datatypes import * >> from version import release >> from request import Request, InvalidRequestError >> from configuration import readSettings, >> ConfigSection, Boolean, dumpSection >> >> >> class ProxyConfig(ConfigSection): >> _dataTypes = {'socket': ControlSocket, >> 'proxyIP': IPAddress, >> 'portRange': PortRange, >> 'TOS': HexNumber, >> 'listen': NetworkAddress, >> 'allow': NetworkRangeList} >> socket = '/var/run/mediaproxy.sock' >> group = 'openser' >> proxyIP = thisHostIP ## defined in utilities >> portRange = (60000, 65000) >> TOS = 0xb8 >> listen = None >> allow = None >> idleTimeout = 60 >> holdTimeout = 60*60 >> forceClose = 0 >> >> >> ## We use this to overwrite some of the settings >> above on a local basis if needed >> readSettings('MediaProxy', ProxyConfig) >> #dumpSection(ProxyConfig) >> >> if ProxyConfig.allow and NetworkRange('Any') not in >> ProxyConfig.allow: >> ## Always add ourselves to the accept list >> myself = NetworkRange('127.0.0.1') >> if myself not in ProxyConfig.allow: >> ProxyConfig.allow.append(myself) >> >> maxDataSize = 8*1024 >> >> try: >> from accounting import accounting, StopRecord, >> StopRecordSerializer, QueuedItemProcessingThread >> except ImportError: >> warning("accounting is enabled but the >> accounting module is missing. accounting is not >> available!") >> accounting = Null() >> dispatcher = Null() >> class StopRecord(object): >> def __new__(typ, *args, **kwargs): >> return None >> else: >> class >> DispatcherNotifyThread(QueuedItemProcessingThread): >> ''' >> Notify the proxydispatcher of calls that did >> timeout and provide >> the dispatcher with the accounting >> information for them. >> ''' >> def __init__(self): >> >> QueuedItemProcessingThread.__init__(self, >> name='DispatcherNotify', >> file='.dispatcher-notify.dat') >> >> def process(self, record): >> address = record['dispatcher'] >> expires = record['expires'] >> del record['dispatcher'], >> record['expires'] >> >> msg = 'timeout %s %s\n' % (record['id'], >> StopRecordSerializer().dump(record)) >> >> s = socket.socket(socket.AF_INET, >> socket.SOCK_STREAM) >> try: >> s.settimeout(5) >> except AttributeError: >> pass >> try: >> start = time.time() >> try: >> s.connect(address) >> s.send(msg) >> result = >> s.recv(maxDataSize).strip() >> except (socket.error, >> socket.gaierror, socket.herror, socket.timeout), >> why: >> try: motive = >> why[1] >> except IndexError: motive = why >> now = time.time() >> if now < expires: >> warning("couldn't notify >> dispatcher at %s (%s). will retry later." % >> (address[0], motive)) >> record['dispatcher'] = >> address >> record['expires'] = expires >> # eventually use a different >> queue only for failures >> delay = 1 >> duration = now - start >> if duration < delay: >> >> time.sleep(delay-duration) >> self.put(record) ## put back >> in the queue and process later >> else: >> error("couldn't notify >> dispatcher at %s (%s). giving up." % (address[0], >> motive)) >> # eventually log the record >> to syslog or a file >> return >> if result.lower() == 'ok': >> info("notified dispatcher at %s >> about expired session %s" % (address[0], >> record['id'])) >> elif result.lower() == 'not found': >> info("session %s already closed >> by the dispatcher at %s" % (record['id'], >> address[0])) >> else: >> error("failed to notify >> dispatcher at %s (returned `%s')." % (address[0], >> result)) >> # eventually log the record to >> syslog or a file >> finally: >> s.close() >> >> def notify(self, session): >> if not session.dispatcher: >> return ## no dispatcher to notify. >> record = StopRecord(session) >> record['dispatcher'] = >> session.dispatcher >> record['expires'] = time.time() + 3600 >> ## try for an hour >> self.put(record) >> >> dispatcher = DispatcherNotifyThread() >> dispatcher.start() >> >> ## Internal state of mediaproxy (DON'T TOUCH!!!) >> >> ## A few shortcuts, for faster access >> proxyIP = ProxyConfig.proxyIP >> minPort = ProxyConfig.portRange[0] >> maxPort = ProxyConfig.portRange[1] >> crtPort = minPort >> >> Sessions = {} >> >> ## For computing traffic through mediaproxy >> byteBucket = [0, 0, 0] >> Traffic = [0, 0, 0] >> >> ## Non public networks. This includes RFC1918 >> networks, localnet (127.0.0.0/8), >> ## broadcast networks (224.0.0.0/4) and 0.0.0.0/8 >> nonPublicNetworks = [ >> {'name': '0.0.0.0', 'value': 0x00000000L, >> 'mask': 0xff000000L}, >> {'name': '10.0.0.0', 'value': 0x0a000000L, >> 'mask': 0xff000000L}, >> {'name': '127.0.0.0', 'value': 0x7f000000L, >> 'mask': 0xff000000L}, >> {'name': '172.16.0.0', 'value': 0xac100000L, >> 'mask': 0xfff00000L}, >> {'name': '192.168.0.0', 'value': 0xc0a80000L, >> 'mask': 0xffff0000L}, >> {'name': '224.0.0.0', 'value': 0xe0000000L, >> 'mask': 0xf0000000L} >> ] >> >> rtpPayloads = { >> > === message truncated === > > -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo.sambucaro(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

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Re: [Serusers] SIP Express Bundle. Step 1.
by Mike Trest - Personal 29 Feb '08

29 Feb '08

At 03:42 AM 2/29/2008, samuel wrote: >I can't recall right now whehter we have discussed the option of >creating in parallel a vmware/xen image so people can start using it >in their favorite virtual solution...more things for the future (?) We discussed. Conclusion was future options , not first out of the gate.

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Re: [OpenSER-Users] [OpenSER-Devel] New module - userblacklist
by Henning Westerholt 29 Feb '08

29 Feb '08

On Tuesday 26 February 2008, Ovidiu Sas wrote: > Can you update the README file with the structure of the table? > Also a "Database setup" section would help a lot. Hi Ovidiu, the README was updated with some informations about the database setup. Please let me know if you have some other suggestions. Cheers, Henning

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Re: [OpenSER-Users] [OpenSER-Devel] Blacklist
by Bogdan-Andrei Iancu 29 Feb '08

29 Feb '08

Yes, it is . Thiago Maluf wrote: > Is it there available this parameter in Openser 1.2.X? > > On 2/29/08, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >> Hi Thiago, >> >> You just have to disable the dns_blacklist support in openser if your >> Asterisk GW is not compliant from reply codes point of view. >> >> Set : >> disable_dns_blacklist = yes >> in your script >> >> Regards >> Bogdan >> >> Thiago Maluf wrote: >> >>> Hi List, >>> In my environment I have one OpenSER and one gateway Asterisk. >>> The Gateway work send the call to PSTN or H.323 World. >>> Then when I send one call to Asterisk and the my E1 link is out, it >>> send to OpenSER one 503 Response! >>> After it, every calll during 4 minutes Openser don't send the call to >>> Asterisk. Then I don't send call to H.323 world for example. >>> I read this page http://www.mail-archive.com/users@openser.org/msg11839.html >>> And I believe that I would possible to change this time to 0 minutes. >>> But I am reading this code section and I don't find where I make this change. >>> Would someone help me? >>> thanks in advanced. >>> Thiago >>> >>> >>> >> > > >

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[OpenSER-Users] New module - userblacklist
by Henning Westerholt 29 Feb '08

29 Feb '08

Hi, i just imported a new module, called 'userblacklist', into the svn trunk. To quote from the README: The userblacklist module allows OpenSER to handle blacklists on a per user basis. This information is stored in a database table, which is queried to decide if the number is blacklisted or not. An additional functionality that this module provides is the ability to handle global blacklists. This lists are loaded on startup into memory, thus providing a better performance then in the userblacklist case. This global lists could also used to prevent the blacklisting of important numbers, as whitelisting is also supported. This is useful for example to prevent the customer from blocking emergency call number or service hotlines. The module exports two functions, check_blacklist and check_user_blacklist for usage in the config file. Furthermore its provide a FIFO function to reload the global blacklist cache. The documentation for this module can be found at: http://www.openser.org/docs/modules/devel/userblacklist.html Best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe

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[OpenSER-Users] OpenSER 1.2.2 and 302 Moved, funny log issue?
by Tobias Lindgren 28 Feb '08

28 Feb '08

Hi there, I was trying to do this in OpenSER 1.2.2: failure_route[1] { if (t_check_status("302")){ xlog("L_ERR", "302 source ip address is $si"); if (src_ip == 192.168.5.1) { # Do something }; }; }; But the proxy sending the 302 (192.168.5.1) is not the one who ends up in my logs, instead my sip-server (192.168.1.1) who initiated the call ends up in the log. "302 source ip address is 192.168.1.1". Have I understood the "$si" psuedo variable wrong? Can I find the correct source address of the 302-message somehow? Br, /Tobias

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[OpenSER-Users] Building openser-1.3.0 on Debian stable
by Sajith T S 28 Feb '08

28 Feb '08

When trying to build openser-1.3.0-tls (tarball from openser.org) on Debian stable, "make all" throws this error: if [ "no" = "yes" ]; then \ cd utils/db_berkeley; make all ; \ fi ; \ /bin/sh: -c: line 4: syntax error: unexpected end of file make: *** [utils] Error 2 The last backslash (after "fi ;") is the problem (line #521 in the top-level Makefile). This is in SVN trunk and 1.3 branch too as far as I can tell. Thanks, Sajith. -- Don't you wish you had more energy... or less ambition? 9DB8FF06 : CB80 0BA6 7D13 B10A 6FBB D43E B4D2 28AD 9DB8 FF06 echo tbkjui(a)opozfo.jo | tr [b-y] [a-x]

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Re: [OpenSER-Users] OpenSER 1.2.2 and 302 Moved, funny log issue?
by Zahid Mehmood 28 Feb '08

28 Feb '08

Thank you for your replies. The sipwise code did have an append branch and a branch route that dropped local branches. -- zahid -----Original Message----- From: "Andreas Granig" <agranig(a)sipwise.com> To: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> Cc: "Zahid Mehmood" <zm23(a)columbia.edu>; "users(a)lists.openser.org" <users(a)lists.openser.org> Sent: 2/28/08 6:32 AM Subject: Re: [OpenSER-Users] OpenSER 1.2.2 and 302 Moved, funny log issue? Hi, Puh, it's been a while since I wrote that, so let me think :) As far as I remember, the idea was to extract the first branch and push it into R-URI, then jump somewhere into the route sections where quite some R-URI manipulation and checking is done (and most of the functions can only operate on r-uri, so...)... after that, the ruri is again appended as branch before sending it out. Only problem is that you can only use one destination. Bogdan, what exactly do you mean by "if it does append_branch somewhere later, it will just duplicate a branch"? Cheers, Andreas Bogdan-Andrei Iancu wrote: > Hi Zahid, > > To be honest, I do not see the purpose of this: > > # get last URI from destination-set and set it as R-URI > avp_delete("$avp(s:tmp)/g"); > $avp(s:tmp) = $ds; > avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); > avp_pushto("$ru", "$avp(s:tmp)"); > > This code fetches the first branch (by using the $ds pv and the regexp) > and push it as RURI; but if there is no following append_branch(), it > has 0 effect. And if it does append_branch somewhere later, it will just > duplicate a branch.... > > Maybe Andreas (here CCed) can give you some details on this..... > > Regards, > Bogdan > > > Zahid Mehmood wrote: >> Hi, >> If get_redirect() is already pushing the contacts as branches, >> then what is the purpose / benefit / drawback of doing something >> like this: >> >> taken from sipwise online config: >> >> if(!get_redirects("1:1")) >> { >> >> xlog("L_ERROR", "Failed to fetch contact '$ct' from >> 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >> acc_db_request("480", "acc"); >> t_reply("480", "Temporarily Unavailable"); >> exit; >> } >> # get last URI from destination-set and set it as R-URI >> avp_delete("$avp(s:tmp)/g"); >> $avp(s:tmp) = $ds; >> avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); >> avp_pushto("$ru", "$avp(s:tmp)"); >> >> >> I'd greatly appreciate your answers as they will help me better >> understand the use of branches/destination set. >> >> Thanks. >> >> >

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Re: [OpenSER-Users] asterisk load balance dispatcher
by Daniel-Constantin Mierla 28 Feb '08

28 Feb '08

Hello, I believe your problem is related nat traversal. Do you do port forwarding on the firewall? As a tip, you should do record routing on openser. To troubleshoot, watch the sip traffic on your network and on the firewall box -- see whether messages are going caller -- asterisk all the time or they are blocked by firewall. Cheers, Daniel On 02/28/08 12:59, Antonio Basti wrote: > > Hi , i’ve a problem with the dispatcher module, for the load balancing > for asterisk > > The schema of the network is this : > > Firewall (public ip: 199.199.199.199:5060) > > | > > | > > ------------------- > > Pc1 pc2 > > Pc1: 192.168.0.1:5060 Asterisk Pc2:192.168.0.3:5060 Asterisk > > Pc1:192.168.0.2:5060 Openser > > The ip public is natted with Openser > > The dispatcher.list is: > > 1 sip:192.168.0.1:5060 > > 1 sip:192.168.0.3:5060 > > Openser.cfg: > > route{ > > if ((method=="INVITE") ){ > > ds_select_dst("1","4"); # 4 = round-robin > > t_relay(); > > exit(); > > } > > } > > In asterisk for test there is an ivr ,and when I make a call from an > other site to this public ip, ast answer but after few seconds I see > from the log of the pc01 the the call hang but from the remote > asterisk (the one where I make the call) the call remain up. > > The configuration of openser is correct ? > > thanks > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 2898 (20080223) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] asterisk load balance dispatcher
by Antonio Basti 28 Feb '08

28 Feb '08

Hi , i've a problem with the dispatcher module, for the load balancing for asterisk The schema of the network is this : Firewall (public ip: 199.199.199.199:5060) | | ------------------- Pc1 pc2 Pc1: 192.168.0.1:5060 Asterisk Pc2:192.168.0.3:5060 Asterisk Pc1:192.168.0.2:5060 Openser The ip public is natted with Openser The dispatcher.list is: 1 sip:192.168.0.1:5060 1 sip:192.168.0.3:5060 Openser.cfg: route{ if ((method=="INVITE") ){ ds_select_dst("1","4"); # 4 = round-robin t_relay(); exit(); } } In asterisk for test there is an ivr ,and when I make a call from an other site to this public ip, ast answer but after few seconds I see from the log of the pc01 the the call hang but from the remote asterisk (the one where I make the call) the call remain up. The configuration of openser is correct ? thanks

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[OpenSER-Users] What is "timezone" field in "subscriber" table for?
by Iñaki Baz Castillo 28 Feb '08

28 Feb '08

Hi, "subscriber" table has a column "timezone". While it could be useful for CDR purposes and so, I find no relationship with OpenSer itself of its modules. Why does this column exist by default in subscriber table? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] What is "timezone" field in "subscriber" table for?
by Bogdan-Andrei Iancu 28 Feb '08

28 Feb '08

Hi Jonathan, More or less this is the whole idea - many openser admins configures a different additional set of fields in the subscriber table, based on the needs of some provisioning tools. Of course, we cannot find a common set for this extra (non-openser) fields, so the most clear way to do it is to keep in the openser default DB struct only the fields that are used by openser. Then, anybody may "customize" the tables, based on his particular needs, without conflicting with the "particular" needs of somebody else ;) Regards, Bogdan Jonathan K. Creasy wrote: > I have applications that use this to display information to the customer (CDR data for example). You may still want to take it out but I just wanted to speak up that there could still be valid use for it. > > -Jonathan > > >> -----Original Message----- >> From: users-bounces(a)lists.openser.org [mailto:users- >> bounces(a)lists.openser.org] On Behalf Of Bogdan-Andrei Iancu >> Sent: Thursday, February 21, 2008 3:50 AM >> To: Henning Westerholt >> Cc: users(a)lists.openser.org >> Subject: Re: [OpenSER-Users] What is "timezone" field in "subscriber" >> table for? >> >> Hi Henning, >> >> I would definitely say so. >> >> Thanks and regards, >> Bogdan >> >> Henning Westerholt wrote: >> >>> On Thursday 21 February 2008, Bogdan-Andrei Iancu wrote: >>> >>> >>>> Hi Inaki, >>>> >>>> Right - this field is more or less used only by serweb (cdr >>>> >> purposes) - >> >>>> openser is not using it at all. >>>> >>>> >>> Hi Bogdan, >>> >>> so i guess its safe if i remove this from trunk? >>> >>> Henning >>> >>> >>> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> > >

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[Serusers] off-topic: nokia e90 sip provisioning via sms ota
by Mark Anthony C. Delfin 28 Feb '08

28 Feb '08

Hi List, Do you have some sample xml config file for provisioning sip settings on nokia E90? I'm using kannel with an smpp connection to send the config. Thanks. Mark D.

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[Serusers] UDP to TCP using SER (as conversion purpose), need ser.cfg example
by Goh Sheng Mai 28 Feb '08

28 Feb '08

hi i have large voip udp packets over internet problems and i need the transfer medium to be tcp. below is what i had in mind. SIPp_client A (TCP out) -> (listen TCP) SER 1 (UDP out) --> Soft-switch (UDP in /out only) --> (to) ring Client B (listen UDP) Client A and Client B registers to above Soft-switch. All the above are individual servers (Linux CentOS based) i need help in a (conversion) ser.cfg. Can anyone post a sample ?? below is my ser.cfg (tried using t_relay_to_udp(), rewritehost(), rewritehostport(), forward functions BUT none seems to work the way i want it to) SER 1 == 192.168.10.203 Softswitch == 192.168.10.251 Client A == 192.168.10.221 or .236 (SIPp client-registered to soft-switch successfully) Client B == 192.168.x.x (registered to soft-switch successfully) Thanks shengmai shengmai_goh(a)innomedia.com.sg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei 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) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #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/print.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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(); #if (method == "INVITE") { # if (uri =~ "sip:*@*"){ # log(1, "Forwarding to Vsp12K\n"); # rewritehostport("192.168.10.204:5060"); # t_relay(); # break; # } #}; if (uri =~ "sip:*@*"){ #log(1, "Forwarding to Vsp12K\n"); #rewritehostport("192.168.10.204:5060"); if (uri =~ "sip:*@192.168.10.203"){ rewritehostport("192.168.10.251:5060"); } t_relay_to_udp("192.168.10.251", "5060"); #t_relay(); break; }; # 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); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # ----------------------------------------------------------------- # Pass on stuff going to PSTN to Asterisk # ----------------------------------------------------------------- #if (uri=~"^sip:*@.*") { # rewritehostport ("*192.168.10.204:5060"); # if (!t_relay()) { # # sl_send_reply ("403", "prout"); # sl_reply_error(); # }; # break; #}; rewritehostport("192.168.10.204:5060"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; break; }

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[OpenSER-Users] No answers whatssoever??
by Taisto Qvist (WM) 27 Feb '08

27 Feb '08

Hi, and sorry if I am getting repetitious but since I am not getting any replies on what I thought was an important question(rfc-complicance) I need to ask the question again...(at least indicate why you disagree with me?) I was hoping someone could comment on what I thought was some fairly clear rfc-references, that indicate that an ACK for 2xx should NOT match the INVITE transaction, since it should be terminated at reception/sending of 2xx. I hope I dont offend in any way, I would just like to get some clarification on this. I hope you can clarify your standpoint, especially if you disagree with me, which I get the feeling you do. Kind Regards Taisto Qvist Taisto Qvist wrote: > I seems like we dont agree on the interpretation of the txn handling > for invite. Sure, there is a timer for handling retransmitted responses, > but thats in earlier states than Terminated. > Sure, there is nothing that says that the inner workings of your > transactions > must exactly behave as the statemachine diagrams in the rfc as long as the > external behavior stays the same, but still, I think that the ACK for 2xx > should not match the INVITE txn. > (It probably boils down to how rfc-isch you want to be.) > > >The end-to-end ACK establish a separate transaction (RFC 3261) and these > >ACKs are not match as part of the INVITE transactions, but correlated > >with them. > > [TQ] > Where does it say that ACKs establish a separate transaction? > There is no defined ACK transaction, only INVITE/nonInvite, server and > client. > The ACK is either a part of the INVITE txn, or its a separate request, > but I would > never call it a standalone transaction. > The real purpose of txn's (in my view), is to enable/simplify forking, > and to > handle retransmissions. Retransmissions of ACK and 2xx, are done by > the UAC/UAS, > so there is no need for a ack-txn. > > >but even describe the wait timer. So, there is no contradiction. > [TQ] > I'd say there is. Where does it describe that this wait-timer should be > used for all responses, or for matching ALL acks? > > My two main reasons for saying that ACKs for 2xx should not be matched, > come from the following two texts. (17.1.1.2, and 17.2.1) > > 17.1.1.2 (client invite txn) > The client transaction MUST be destroyed the instant it enters the > > "Terminated" state. This is actually necessary to guarantee correct > operation. The reason is that 2xx responses to an INVITE are treated > differently; each one is forwarded by proxies, and the ACK handling > in a UAC is different. Thus, each 2xx needs to be passed to a proxy > core (so that it can be forwarded) and to a UAC core (so it can be > acknowledged). No transaction layer processing takes place. > Whenever a response is received by the transport, if the transport > layer finds no matching client transaction (using the rules of > Section 17.1.3), the response is passed directly to the core. Since > the matching client transaction is destroyed by the first 2xx, > > subsequent 2xx will find no match and therefore be passed to the > core. > > [TQ] > It even says "actually nessesary to guarantee...." > Since the txn's are there to (among other things) absorb retransmissions, > the receiving the 2xx MUST destroy the txn, so that when the next > retransmission > of 2xx is received, it is not consumed by the txn layer. > This is what your txn-layer does for 3++ right? Any additional 3++ > received on > a txn while waiting for Timer D to expire, will just be consumed, and > the proxy > core will never have to process it. > > > 17.2.1. (server invite txn) > The purpose of the "Confirmed" state is to absorb any additional ACK > messages that arrive, triggered from retransmissions of the final > response. > ..snip... > Once the transaction is in the "Terminated" state, it MUST be destroyed > immediately. > > [TQ] > The Confirmed state is there to absorb retransmissions, not the > terminated. > When receiving 2xx you go directly to terminated bypassing confirmed. > > Also, in 18.2.1 the text explicitly says: > > .. Note that when a UAS core sends a 2xx response to INVITE, > the server transaction is destroyed. This means that when the ACK > arrives, > there will be no matching server transaction, and based on this rule, > the ACK is passed to the UAS core, where it is processed. > > Thats faily clear right? :-) > The think that puzzles my a little bit though, is that you must have added > "extra" code for this matching...? > I mean, you're using the ;branch=z9xxxx for transaction matching right, if > its there? And the ACK for 2xx doesnt have the same branch, indicating > a "new txn". > > So matching ALL ACKs would imply that for any ACK that doesnt match a txn, > your really checking up Call-Id's, tags and stuff, just to be able to > match > this ACK to the 2xx? > > Or are you checking callid's, cseq's, tags, and stuff, for ALL > requests in your > txn matching? > > Now, I been working on a sip-stack the last few years myself, and I > naturally > know that there is often a need to match the ACK, and the stack I've > build has > similar functions, it just doesnt do it on the txn layer, where we try > to be as > RFC-isch as possible. > > What I would like, is simply have some function-parameters to the > t_check_trans(<DEFANGED_param>), > or even better in my little mind, a "modparam("tm", > "ack_2xx_matching", 1)". > > I am looking forward to hearing your reply, and in the mean time, > thanks to all of you developers for an extraordinary software :-) > > Regards > Taisto Qvist > > > Hi guys, > > > > Just to bring some clarifications on the TM module. > > > > once a transaction is completed (negative or 2xx reply), it is put on > > wait (wait timer - see RFC3261) for catching any potential > > retransmissions of replies. > > So, the transaction is completed, but not removed from memory - RFC does > > not say that you need to trash immediately all the transaction > > information, but even describe the wait timer. So, there is no > > contradiction. > > > > The ACK (for 2xx)catching is done based on the completed INVITE > > transaction (from wait timer) - nothing else. > > > > > > Inaki, could you please detail what you mean by: > > > > <quote> > > In my opinion OpenSer does a special treatment for ACK in tm mode, > even if > > they are for failed transaction (hop-by-hop ACK's) or succesfull INVITE > > (end-to-end ACK's). > > > > </quote> > > > > Maybe I can explain you more if I understand you question.... > > > > Regards, > > Bogdan > ---- Taisto Qvist, IP-Solutions Mobile: +46-708-88 02 63 "We are Pentium of Borg. Division is futile, You will be approximated"

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[OpenSER-Users] no-answer timeout
by Mark Sayer 27 Feb '08

27 Feb '08

Hi - I am using 1.2.1-notls. When using t_relay() to INVITE a call, I assume that the timeout before returning a 408 would be governed by fr_inv_timer and fr_inv_timer_avp. Although the 1.2 documentation states that the default for fr_inv_timer is 120 seconds I seem to get a timeout at 60 seconds and neither fr_inv_timer or fr_inv_timer_avp seems to have any effect. Have a made an error in interpreting the docs? How should I be varying this timeout? Thanks, Mark

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[OpenSER-Users] Out of order notifications
by Leonard Broman 27 Feb '08

27 Feb '08

Hi! I have integrated OpenSER as a dedicated presence server in an ims network and I'm somewhat suffering from the delays the traversal of the ims core causes. Here's whats happening: Upon subscription the presence module sends a 202 Accepted (actually named OK, cosmetic bug?). At almost the same instant it also sends the first, in-dialog notify. What happens later is that sometimes the notify arrives before the ok. When this happens the client (on top of jain-sip) cannot identify the subscription dialog for the notification. It does not treat it correctly and does not respond OK to it. The absence of the response results in a timeout and the presence server invalidates the subscription. Now, to solve this issue, would it be wise to introduce a parameter in the presence module which delays to transmission of the first notification to ease up the situation for the client. Another solution is of course that the client simply inspects the local tag to prematurely identify the upcoming sip dialog for the subscription and then be able to treat the request properly. However, how does this stand against the standards? I would be grateful for your input on the subject. Leonard Broman

1 0

[OpenSER-Users] XMLRPC
by Helmut Kuper 27 Feb '08

27 Feb '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to get some Informations from openser via mi_xmlrpc module. Unfortunately I allways get an "404 Not found" Reply back from openser. I checked the communication between openser and php xml client. My client sends this to openser: POST / HTTP/1.0 Host: 85.16.246.9 Content-Type: text/xml User-Agent: me Content-length: 157 <?xml version="1.0"?> <methodCall> <methodName>get_statistics</methodName> <params> <param><value><string>tm:</string></value></param> </params></methodCall> That looks like as documentated in mi_xmlrpc module doc. I send different MI commands to openser: real and non real. But I always get that 404 error. In mi_xmlrpc log is nothing except those 404 answeres again. Debug level 9 says nothing. Sending MI commands to openser via mi_fifo works. Is there anything else to configure for xmlrpc? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHxEEz4tZeNddg3dwRAmpwAJoDKWW6bWU3zRyaaisa6HOOX6JdKwCcDAqM 2pmp4F96L85s8SmEI5L76ak= =23Fr -----END PGP SIGNATURE-----

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Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA
by Anca Vamanu 27 Feb '08

27 Feb '08

Hi, I made the fix for both 1.3 and 1.4 versions. There is no directory with patches. The newest version is found in the on svn for branch 1.3 and trunk respectively. If you want to make use of this patch you have to update to download the latest svn version. See: http://www.openser.org/mos/view/Download/. <http://www.openser.org/mos/view/Download/> regards, Anca Vamanu Jeremy George wrote: > Anca, > > Is this a patch for 1.3? Or, a fix that went into the 1.4 development > tree? If it's for 1.3, how do I get it? I looked through openser.org but > can't find a directory of patches. Thanks much. > > - Jeremy > > > >> -----Original Message----- >> From: users-bounces(a)lists.openser.org [mailto:users- >> bounces(a)lists.openser.org] On Behalf Of Anca Vamanu >> Sent: Thursday, February 21, 2008 11:14 AM >> To: Zahid Mehmood >> Cc: users(a)lists.openser.org >> Subject: Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA >> >> Hi, >> >> Thank you for the print. It helped finding the bug. >> I have just committed a fix. It would be great if you would update with >> the svn version, test and confirm if it works. >> >> Thanks, >> Anca Vamanu >> >> Zahid Mehmood wrote: >> >>> Hi Anca, >>> Thank you for your quick response. Attached below is the >>> information you requested. >>> >>> Thanks. >>> >>> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> > > > >

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Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA
by Jeremy George 27 Feb '08

27 Feb '08

Hey Anca, That's great. Thanks. So, I have now openser-1.3.x-svn-latest.tgz . If it doesn't wear your patience thin with too-simple questions, I'd like to ask just a couple more. Is this version still considered a stable release? And, I can't find a file within the tarball that distinguishes this version from others. For example, the ChangeLog doesn't show it. Where would I find such information? Again, thanks for your expertise, and patience. - Jeremy >-----Original Message----- >From: Anca Vamanu [mailto:anca@voice-system.ro] >Sent: Wednesday, February 27, 2008 10:24 AM >To: 'Jeremy George' >Cc: users(a)lists.openser.org >Subject: Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA > >Hi, > >I made the fix for both 1.3 and 1.4 versions. There is no directory with >patches. The newest version is found in the on svn for branch 1.3 and >trunk respectively. If you want to make use of this patch you have to >update to download the latest svn version. See: >http://www.openser.org/mos/view/Download/. ><http://www.openser.org/mos/view/Download/> > >regards, >Anca Vamanu > >Jeremy George wrote: >> Anca, >> >> Is this a patch for 1.3? Or, a fix that went into the 1.4 development >> tree? If it's for 1.3, how do I get it? I looked through openser.org but >> can't find a directory of patches. Thanks much. >> >> - Jeremy >> >> >> >>> -----Original Message----- >>> From: users-bounces(a)lists.openser.org [mailto:users- >>> bounces(a)lists.openser.org] On Behalf Of Anca Vamanu >>> Sent: Thursday, February 21, 2008 11:14 AM >>> To: Zahid Mehmood >>> Cc: users(a)lists.openser.org >>> Subject: Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA >>> >>> Hi, >>> >>> Thank you for the print. It helped finding the bug. >>> I have just committed a fix. It would be great if you would update with >>> the svn version, test and confirm if it works. >>> >>> Thanks, >>> Anca Vamanu >>> >>> Zahid Mehmood wrote: >>> >>>> Hi Anca, >>>> Thank you for your quick response. Attached below is the >>>> information you requested. >>>> >>>> Thanks. >>>> >>>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.openser.org >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> > >

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Re: [OpenSER-Users] Call timeout
by Bogdan-Andrei Iancu 27 Feb '08

27 Feb '08

Hi David, The setup time is actually controlled by TM (during the INVITE transaction). If you want to set terminate a call, see the dialog module, in openser 1.3 version: http://www.openser.org/docs/modules/1.3.x/dialog.html#AEN461 regards, Bogdan David Villasmil wrote: > Hello, > > Timeout for a setup I understand is done with SST/Dialog, I mean > disconnect an established call. > > > thanks > > On Wed, Feb 27, 2008 at 10:16 AM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > Hi David, > > What do you mean by "call timeout" - do disconnect an established call > after some time? or timeout during call setup? > > Regards, > Bogdan > > David Villasmil wrote: > > Hello all, > > > > I have a network already operational for voip-voip/voip-pstn > > servicing about 20k customer and all is "well". At the moment I'm > > using Asterisk as a B2BUA mainly for call timeout capabilities. Is > > there absolutely NO WAY of setting a call timeout (disconnect) > after X > > seconds? Do I have to use a separate B2BUA? > > > > > > Thanks everyone. > > > > David > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

1 0

[Serusers] OpenSER summit at VONX Spring 2008
by Greger Viken Teigre 27 Feb '08

27 Feb '08

I know many SER users also use OpenSER in combination with SER. Indeed, I see also quite a lot of confusion with people asking OpenSER questions on the SER mailing list... So, not to add to the confusion: OpenSER is a fork of the original SIP Express Router project, where OpenSER and SER code bases started to diverge at that point. It was only the SIP proxy of iptel.org that was forked, thus: SEMS, SERWeb, and sipsak are still iptel.org only projects with no forks (rtpproxy.org, formerly loosely coupled with iptel.org project has now its own home: http://rtpproxy.org). So, with that clarification, some seruser subscribers may find the below announcement of interest. I forward it on the request of Bogdan, one of the maintainers of OpenSER. g-) The second edition of OpenSER Summit will take place in San Jose, USA , on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of the event, see http://www.openser.org/mos/view/OpenSER-Summit-2008. All participants to register via OpenSER site before Friday, March 7th, will get free access to the OpenSER event. This OpenSER Summit edition is sponsored by a parallel OpenSER related event, the OpenSER Pavilion . The pavilion is a common exhibiting area - booth 1027, inside VoN expo, gathering, under the OpenSER name, six different companies working or using the project. You can find more about the OpenSER Pavilion and its participants here - http://www.openser.org/mos/view/OpenSER-Pavilion-2008. The dual event, OpenSER Summit & Pavilion is new concept of a more complex event, aiming to create a larger diversity and to give more power to the understanding of the OpenSER project.

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[OpenSER-Users] User Documentation - request for participation
by Henning Westerholt 27 Feb '08

27 Feb '08

Hello all, its a well know fact that there exists some issues with the OpenSER documentation, for some parts there is almost no documentation available at all. Existing documentation is sometimes a little bit unstructured or incomplete. Its hard for new users to understand the configuration of the server, let alone how they should setup some services. And after he or she has gone through the basics, its again the same for configuring complex modules.. During the last IRC meeting we decided to start a new effort to tackle this problems more efficiently than in the past. So a new mailling list 'openser-docs' has been created, targeted in improving and extending the existing documentation, and also create new content, for example a 'OpenSER Guide'. As its difficult for many of the existing developers to write user related documentation, because they're already to deep involved into the project, i would like to ask here for participation from you. We're seeking people that are also dissatisfied with the current situation, and want invest something to improve this. No coding experiences or knowledge about certain tools or languages are necessary. So please join us and subscribe at http://lists.openser.org/cgi-bin/mailman/listinfo/openser-docs ! Thank you, Henning

1 0

[OpenSER-Users] Call timeout
by David Villasmil 27 Feb '08

27 Feb '08

Hello all, I have a network already operational for voip-voip/voip-pstn servicing about 20k customer and all is "well". At the moment I'm using Asterisk as a B2BUA mainly for call timeout capabilities. Is there absolutely NO WAY of setting a call timeout (disconnect) after X seconds? Do I have to use a separate B2BUA? Thanks everyone. David

2 1

[OpenSER-Users] Fwd: [Serdev] RTPproxy has gotten its own mailing lists
by Ovidiu Sas 27 Feb '08

27 Feb '08

FYI Regards Ovidiu Sas ---------- Forwarded message ---------- From: Maxim Sobolev <sobomax(a)sippysoft.com> Date: Tue, Feb 26, 2008 at 7:17 PM Subject: [Serdev] RTPproxy has gotten its own mailing lists To: serdev(a)lists.iptel.org Hi, Just a quick note that today we've finally launched mailing lists for the RTPproxy. Anybody is welcome to join. http://lists.rtpproxy.org/mailman/listinfo Regards, -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts T/F: +1-646-651-1110 Web: http://www.sippysoft.com _______________________________________________ Serdev mailing list Serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

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[OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA
by Zahid Mehmood 26 Feb '08

26 Feb '08

Hi All, I need help with troubleshooting openser 1.3.0 crash. I am running openser 1.3.0 (not from svn) to provide only BLA presence service. The config file is based on the sample provided on openser website. This is the first time openser actually crashed. Twice before it stopped working but the process never died. Originally openser was running with debug=3. I have changed it to log more information now. What else can i do to generate more useful information to help resolve this issue? Thanks in advance for your help! -- Zahid From openser log: Feb 15 16:56:52 torte /usr/sbin/openser[25542]: 7bfbf097-25539(a)10.0.0.129 : handle_publish Feb 15 16:56:52 torte /usr/sbin/openser[25539]: 15b7f2f2-11174(a)10.0.0.129 : bla_handle_notify OK Feb 15 16:57:13 torte /usr/sbin/openser[25570]: CRITICAL:core:receive_fd: EOF on 17 Feb 15 16:57:13 torte /usr/sbin/openser[25522]: INFO:core:handle_sigs: child process 25551 exited by a signal 11 Feb 15 16:57:13 torte /usr/sbin/openser[25522]: INFO:core:handle_sigs: core was generated Feb 15 16:57:13 torte /usr/sbin/openser[25522]: INFO:core:handle_sigs: terminating due to SIGCHLD Feb 15 16:57:13 torte /usr/sbin/openser[25534]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25560]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25568]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25565]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25533]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25531]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25553]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25570]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25538]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25558]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25542]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25539]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25548]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25540]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25522]: NOTICE:presence:destroy: destroy module ... bash-3.00# openser -V version: openser 1.3.0-notls (i386/linux) 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, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 3304 2007-12-11 12:55:11Z bogdan_iancu $ main.c compiled on 21:52:01 Jan 6 2008 with gcc 3.4.6 gdb info: Core was generated by `/usr/sbin/openser'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/libdl.so.2...done. (gdb) backtrace #0 0x00b3bba3 in strncmp () from /lib/tls/libc.so.6 #1 0x00140bcb in get_dialog (dialog=0xb60fe518, hash_code=22) at hash.c:289 #2 0x0014ae10 in subs_cback_func (t=0xb61210b0, cb_type=256, ps=0xdb6314) at send_subscribe.c:248 #3 0x00d945b2 in run_trans_callbacks (type=256, trans=0xb61210b0, req=0x0, rpl=0xffffffff, code=408) at t_hooks.c:205 #4 0x00da14ca in local_reply (t=0xb61210b0, p_msg=0xffffffff, branch=0, msg_status=408, cancel_bitmap=0xbfe95834) at t_reply.c:1255 #5 0x00da5f4c in timer_routine (ticks=89623, attr=0x0) at timer.c:254 #6 0x080b403f in start_timer_processes () at timer.c:275 #7 0x08066d60 in main (argc=1, argv=0xbfe95a94) at main.c:873

3 6

[OpenSER-Users] MI commands
by David Villasmil 26 Feb '08

26 Feb '08

Hello all, Is there a way to generate a BYE message, disconnect message via Manager Interface? thanks all David

2 1

[OpenSER-Users] openser+opendiameter integration--->kindly pay attention
by bhargav.b＠renovau.com 26 Feb '08

26 Feb '08

hi all, i am trying to authenticate a user using openser and opendiameter.currently i have configured both of them on the same pc.my problem is that i cannot use anything other than opendiameter due to some architechture restrictions of our company plan. when i run xlite[softphone],openser,opendiameter i get error in opendiameter client saying that"missing session id avp".i am not able to identify where exactly the problem lies,either in openser's auth_diameter module which is developed for diameter or in the opendiameter. if somebody have done anything like what i am trying to do or if anybody have some idea please help me out.Because i think its really very hard to do this without somebodies help

1 0

Re: [OpenSER-Users] OpenSER failure when MySQL redundant database fails
by Henning Westerholt 26 Feb '08

26 Feb '08

On Monday 25 February 2008, Arturo Díaz Almagro wrote: > Please, see my comments below. > > [..] > > For recent mysql driver versions this is not needed anymore, i've > > committed in revision 3606 a improvement to the trunk version of openser > > that uses the recently introduced extensions in the mysql lib. This could > > be easily backported to your version, take a look into the mysql > > documentation if this different timeouts are supported on your driver > > version. > > Does that mean that I could compile the mysql module in the trunk version > and replace the old mysql.so module by the new generated one from the > trunk? Is it fully backward compatible? > > Kind regards. Hi Arturo Díaz Almagro, (lets continue this discussion on the user list) No, but this will not work at all. Even replacing the code and recompile will not work, the changes were to extensive. You need to backport the changes from the mentioned revision to the driver you use internally in your company. You could get the changes with 'svn diff' or search on the devel list for the commit message. Cheers, Henning

2 2

[Serusers] OPTION using sipsak
by YAHYAOUI Mohammed 26 Feb '08

26 Feb '08

Hello, Having 1 boxes with SIP service Box A: ip a.b.c.d with sip at udp 5060 and a NMS machine with ip x.y.z.t I am looking to monitor the SIP service via OPTION sent from NMS (x.y.z.t) to the box A (no user defined on the box) Any one to help how to write the OPTION request using sipsak I tried this way but no success sipsak -f option -s sip:a.b.c.d (I don't know if we have to set the udp port with sipsak) On file option I have: OPTIONS sip:a.b.c.d:5060 SIP/2.0 Max-Forwards: 70 To: <sip:ping@a.b.c.d:5080> From: NMS <sip:x.y.z.t> Call-ID: 83185420572 CSeq: 8764 OPTIONS Allow: ACK,CANCEL Accept: application/sdp Content-Length: 0 BR

2 1

[OpenSER-Users] rtpproxy on separate server
by Ron 26 Feb '08

26 Feb '08

Hi, What is the advantage of having rtpproxy on a separate server against having it on the same server as the sip registrar/proxy? TIA Regards, Ron

3 2

Re: [Serusers] Intercept SIP call
by roayu 26 Feb '08

26 Feb '08

Thanks for the explanation! I will try to work it out. Roa :D -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Monday, February 25, 2008 8:11 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Intercept SIP call Ringing multiple clients is something the server does -- not the client. Eg: User A logs into SER with IP:22.33.44.55 and leaves his client running. He then goes to work and logs into SER with IP: 88.99.66.77 In SER, there's now a table that has that information: User A: 22.33.44.55, 88.99.66.77 When user B calls, SER rings every place User A is logged in from. The first phone to answer gets the call. Call pickup is something completely different. And yes, for that, you'd need Asterisk running. N. roayu wrote: > Thanks. > > So, if the client aren't able to perform this function. Can SER do that? Or > I need to setup the Asterisk server to help me to perform the 'call pickup' > functions. Correct me if I'm wrong. > > Thanks. > > Roa Yu > > -----Original Message----- > From: SIP [mailto:sip@arcdiv.com] > Sent: Monday, February 25, 2008 11:17 AM > To: roayu > Cc: samuel; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Intercept SIP call > > When your client registers to SER as a particular user, that > registration is stored in SER. > > If you register multiple times with the same user, ALL of those contacts > are called when a call is sent to that user. > > N. > > > roayu wrote: > >> Thanks for your reply and explanation, samuel. >> >> But what do you mean by "If you want to send the initial call to both >> phone B and C, either in parallel or sequentally, then SER can easily >> do the job"? Could you please further explain on this or you can >> provide me so links to read bout this. >> >> Thanks! >> >> Roa Yu. >> >> ------------------------------------------------------------------------ >> >> *From:* samuel [mailto:samu60@gmail.com] >> *Sent:* Friday, February 22, 2008 4:39 PM >> *To:* roayu >> *Cc:* serusers(a)lists.iptel.org >> *Subject:* Re: [Serusers] Intercept SIP call >> >> The way you have explained the use case would require the phones to do >> complex SIP operations. >> Phone C need to know the dialog identifiers (dialog info package >> probably would help)of the call between A and B and then use replaces >> header to "intercept the call". I presonally haven't seen phones doing >> this. >> >> If you want to send the initial call to both phone B and C, either in >> parallel or sequentally, then SER can easily do the job. Just read a >> few mails or the getting started document and you'll see how to do this. >> >> Hope it helps, >> Samuel. >> >> 2008/2/20, roayu <roayu(a)ctisys.net <mailto:roayu@ctisys.net>>: >> >> Hi there! >> >> Recently, I had setup a SIP environment with few sip phones, SIPphone >> A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to >> SIPphone B, but B is not able to answer the call. Can SIPphone C >> answer A's call? >> >> I would like to know whether is SER able to have the intercept sip >> calls function? If yes, what module or configuration that I need to set? >> >> Thanks in advance! >> >> Best regards, >> >> RoaYu >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> >> http://lists.iptel.org/mailman/listinfo/serusers >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >>

1 0

[Serusers] gw table lookup 2.0
by Mike Trest - Personal 26 Feb '08

26 Feb '08

Hi, can someone point me to a 2.0 config snip that will lookup gw_name and return data element[s] from "gw" table? As an alternative, a simple found/not found return value will do. ..mike..

1 1

[OpenSER-Users] Add record route in reply route?
by Peter P GMX 26 Feb '08

26 Feb '08

I have the following scenario: UA - OpenSER - SIPX - MicrosoftUM2007 As OpenSER has the public IP address and SIPX has no public IP, Openser has to be in the loop of all SIP messages. Therefore I set a record-route header in OpenSER. SipX is needed for the UDP/TCP gateway to MS UM2007 so therefore SipX needs to add another record-route header. UM2007 then receives a record with 2 record-routes (the one of OpenSER and the one of SipX) Unfortunately our UM2007 replaces the OpenSER record-route in the OK messages by the IP of SipX IP only, so Openser is out of the loop. The UA then tries to reply with ACK to the (private) IP of SipX and fails. My question: Ist here a chance to add another record-route with OpenSer's IP in OpenSER 1.3 in the reply route? According to the documentation in rr it doesn't work. Any hint is appreciated. Best regards Peter

2 1

[OpenSER-Users] email address update
by Daniel-Constantin Mierla 25 Feb '08

25 Feb '08

Hello, my new email address for any matter is miconda(a)gmail.com -- as I was not able to access my inbox over last week, please resend to the new address any message you sent to me (not the ones sent to the lists, I will pick them up from there). Thank you, Daniel

2 1

[OpenSER-Users] #remove
by Silvio Netto 25 Feb '08

25 Feb '08

_________________________________________________________________ Conheça o Windows Live Spaces, a rede de relacionamentos do Messenger! http://www.amigosdomessenger.com.br/

1 0

[OpenSER-Users] Status code 408 returned by OpenSER when fr_timer expires; should it be 504?
by Anatoly Pidruchny 25 Feb '08

25 Feb '08

Hello everybody, I noticed that OpenSER replies with status code "408 Request Timeout" when the fr_timer (tm module) expires. I am wondering if the response code "504 Server Timeout" would be more appropriate in this case. The status code 408 means that "the server could not produce a response, e.g., a user location, within the time indicated in the Expires request-header field. The client MAY repeat the request without modifications at any later time."; and the status code 504 means "the server did not receive a timely response from an external server it accessed in attempting to process the request." It sounds to me that the description of the status code 504 better describes what happens in OpenSER when the fr_timer expires. Any comments? I am using OpenSER 1.2.x. Best regards, Anatoly Pidruchny.

2 2

Re: [OpenSER-Users] OpenSER failure when MySQL redundant database fails
by Henning Westerholt 25 Feb '08

25 Feb '08

On Friday 22 February 2008, Arturo Díaz Almagro wrote: (forward to user list for the archives) > This is my case. We have a SIP telephony system based on OpenSER with a > redundant MySQL database ring, so the OpenSER queries always to the primary > DB unless it fails. When the "master" database dies, a monit process > detects the failure and modify properly the database entry in the DNS zone > file. We have tested this redundancy in two way I expose you here: > > - mysql process is killed in the database host machine. In this case > OpenSER realizes the failure and asks the DNS inmediately for the > database IP address and the service keeps working as expected. No problem > here. > > - shutdown the interface of the machine where mysql is hosted. In this > case OpenSER does not ask the DNS for the new address. It keeps trying > over and over the connection with the master and until the interface is not > turn on the system keeps hung. > > The difference between the two cases is that in the first case, the host > machine responses with an ICMP (port destination unreachable) when OpenSER > queries the database. In the second case we have no answer from the MySQL > host machine. > > So, if the mysql's host interface is shutdown the system hangs but if the > mysql process dies the system works properly. In the first case both > machines (OpenSER and MySQL) has IP visibility, not in the second, so the > problem should be related to that. Do you know why this is happening? Might > you, please, be so kind to tell me how to fix this problem or at list how > to take a workaround to get it working properly? Hi Arturo, Well, this problem is probably caused because the mysql_query function works synchronly, i.e. it blocks the process if no progress on the query could be made. You could easily reproduce this with simple IP filtering with iptables. If you don't get a negative reply to the query, it will simply hang for the TCP timeout from the kernel. We've written in the past a custom extension to the mysql module, called mysql-mt. This elevates this problem by uses a multi-threading approach. But this worked not that great. For recent mysql driver versions this is not needed anymore, i've committed in revision 3606 a improvement to the trunk version of openser that uses the recently introduced extensions in the mysql lib. This could be easily backported to your version. Take a look into the mysql documentation if this different timeouts are supported on your driver version. Cheers, Henning

1 0

[OpenSER-Users] UA Gets Disconnected
by Ali Jawad 25 Feb '08

25 Feb '08

Hi All I am using 1.3 with tls enabled, I am using the default certs that come with the source code and I did disable all the tls_verify options, I can logon to my UA and make calls however there is one problem, every couple minutes the sip client disconnects and re-connects "it does have a re-login option". Any ideas why the UA is getting disconnected every couple of minutes ? Note: I have also tried with UDP without tls and the same thing happened upon monitoring SIP packets I noticed the following, as long as the client is connected it regularly sends REGISTER packets, when the disconnect happens the following happens SIP/2.0 401 Unauthorized Thx With Regards Ali Jawad

1 0

[OpenSER-Users] ERROR:core:_tls_read: something wrong in SSL:
by John Barry 25 Feb '08

25 Feb '08

Hello, I am trying to do a performance test of OpenSER (1.3.0) with TLS using SIPP as the UAS/UAC. The TLS authentication seems to work and the calls are stablished and then terminated (SIPP default scenario). However, I am receiving the following error message from OpenSER: Feb 19 12:58:40 [12709] ERROR:core:_tls_read: something wrong in SSL: Feb 19 12:58:40 [12709] ERROR:core:tcp_read_req: failed to read What do these messages mean? Can I ignored them? The SIP command I am using is: sipp -s 100 -sn uac 192.168.1.10:5061 -t ln -tls_cert 101-cert.pem -tls_key 101-privkey.pem The TLS configuration I am using in my openser.cfg file is: disable_tls = no listen = tls:192.168.1.10:5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = SSLv23 tls_certificate = "/openser/etc/openser/tls/proxy/proxy-cert.pem" tls_private_key = "/openser/etc/openser/tls/proxy/proxy-privkey.pem" tls_ca_list = "/openser/etc/openser/tls/proxy/proxy-calist.pem" I will appreciate any help about this matter. Thanks, JB _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

3 3

Re: [Serusers] Intercept SIP call
by roayu 25 Feb '08

25 Feb '08

Thanks. So, if the client aren't able to perform this function. Can SER do that? Or I need to setup the Asterisk server to help me to perform the 'call pickup' functions. Correct me if I'm wrong. Thanks. Roa Yu -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Monday, February 25, 2008 11:17 AM To: roayu Cc: samuel; serusers(a)lists.iptel.org Subject: Re: [Serusers] Intercept SIP call When your client registers to SER as a particular user, that registration is stored in SER. If you register multiple times with the same user, ALL of those contacts are called when a call is sent to that user. N. roayu wrote: > > Thanks for your reply and explanation, samuel. > > But what do you mean by "If you want to send the initial call to both > phone B and C, either in parallel or sequentally, then SER can easily > do the job"? Could you please further explain on this or you can > provide me so links to read bout this. > > Thanks! > > Roa Yu. > > ------------------------------------------------------------------------ > > *From:* samuel [mailto:samu60@gmail.com] > *Sent:* Friday, February 22, 2008 4:39 PM > *To:* roayu > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Intercept SIP call > > The way you have explained the use case would require the phones to do > complex SIP operations. > Phone C need to know the dialog identifiers (dialog info package > probably would help)of the call between A and B and then use replaces > header to "intercept the call". I presonally haven't seen phones doing > this. > > If you want to send the initial call to both phone B and C, either in > parallel or sequentally, then SER can easily do the job. Just read a > few mails or the getting started document and you'll see how to do this. > > Hope it helps, > Samuel. > > 2008/2/20, roayu <roayu(a)ctisys.net <mailto:roayu@ctisys.net>>: > > Hi there! > > Recently, I had setup a SIP environment with few sip phones, SIPphone > A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to > SIPphone B, but B is not able to answer the call. Can SIPphone C > answer A's call? > > I would like to know whether is SER able to have the intercept sip > calls function? If yes, what module or configuration that I need to set? > > Thanks in advance! > > Best regards, > > RoaYu > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] Fwd: SIP Express Bundle ready-to-go idea, request for input
by Mike Trest - Personal 25 Feb '08

25 Feb '08

Greger, et. al., I like your idea enough to volunteer as a tester + documentation writer. I do a lot of tech and user writing. Active in VoIP since the beginning. I am active VoIP engineer and developer. Also PSTN & VoIP network designer-builder. I have my own private machines, network, and VoIP gateway resources. Prefer CentOS or FC* ..mike.. >Hi guys, >I have been playing with the following idea: >Create a ready-to-run OS image with everything that is needed for >iptel.org apps pre-installed + a complete installation of: >* SER 2.0 (release) >* rtpproxy >* SEMS >* SERweb >* maybe sipsak, some monitoring tools, etc > > >The idea is to create a small script 'config_iptelorg' for >configuring the installation to your needs. You should then be able >to download the ready image, boot it, go through the script and have >an up and running iptel.org proxy and app server just like the >iptel.org free SIP service in maybe 10-15 minutes. This way you >could host a SIP service for your own domain with close to no setup at all. > >Some questions to you: >* Is there any interest for this at all? >* I was thinking about using Ubuntu 7.10 server as the OS. Any >thoughts/preferences? >* Should the image be an Amazon EC3 image (you could use >http://www.rightscale.com and get it running in no time with 10 >run-hours free) or should it be a VMware appliance to be run with >free VMware Player? >* Other suggestions/comments? > >I could need some help with this, anyone interested in lending me a >hand? (could be anything, documentation, testing, installation, etc) >g-)

6 6

[Serusers] Openser - Ejabberd Integration
by Frank Gonzalez 25 Feb '08

25 Feb '08

Hi, im trying to stablish the real time connection between Openser and Ejabberd both with mysql support, any suggestions ? -- Thanks Frank Gonzalez 414-6260492

2 1

Re: [Serusers] Intercept SIP call
by roayu 25 Feb '08

25 Feb '08

Thanks for your reply and explanation, samuel. But what do you mean by "If you want to send the initial call to both phone B and C, either in parallel or sequentally, then SER can easily do the job"? Could you please further explain on this or you can provide me so links to read bout this. Thanks! Roa Yu. _____ From: samuel [mailto:samu60@gmail.com] Sent: Friday, February 22, 2008 4:39 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Intercept SIP call The way you have explained the use case would require the phones to do complex SIP operations. Phone C need to know the dialog identifiers (dialog info package probably would help)of the call between A and B and then use replaces header to "intercept the call". I presonally haven't seen phones doing this. If you want to send the initial call to both phone B and C, either in parallel or sequentally, then SER can easily do the job. Just read a few mails or the getting started document and you'll see how to do this. Hope it helps, Samuel. 2008/2/20, roayu <roayu(a)ctisys.net <mailto:roayu@ctisys.net> >: Hi there! Recently, I had setup a SIP environment with few sip phones, SIPphone A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to SIPphone B, but B is not able to answer the call. Can SIPphone C answer A's call? I would like to know whether is SER able to have the intercept sip calls function? If yes, what module or configuration that I need to set? Thanks in advance! Best regards, RoaYu _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers>

2 1

Re: [Serusers] FW: Intercept SIP call
by roayu 25 Feb '08

25 Feb '08

Thanks for your reply. This means that I need to setup another SIP server (Asterisk) to help me to handle the 'call pickup', right? Thanks. RoaYu -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Friday, February 22, 2008 8:39 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] FW: Intercept SIP call This is called 'call pickup' in the PBX world, and most PBX software can handle it (see Asterisk). N. roayu wrote: > > Resend. > > > > ------------------------------------------------------------------------ > > *From:* roayu > *Sent:* Wednesday, February 20, 2008 11:21 AM > *To:* serusers(a)lists.iptel.org > *Cc:* roayu > *Subject:* Intercept SIP call > > > > Hi there! > > > > Recently, I had setup a SIP environment with few sip phones, SIPphone > A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to > SIPphone B, but B is not able to answer the call. Can SIPphone C > answer A's call? > > > > I would like to know whether is SER able to have the intercept sip > calls function? If yes, what module or configuration that I need to set? > > > > Thanks in advance! > > > > > > Best regards, > > RoaYu > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] UDP to TCP using SER (as conversion purpose), need ser.cfg example
by Goh Sheng Mai 25 Feb '08

25 Feb '08

hi i have large voip udp packets over internet problems and i need the transfer medium to be tcp. below is what i had in mind. SIPp_client A (TCP out) -> (listen TCP) SER 1 (UDP out) --> Soft-switch (UDP in /out only) --> (to) ring Client B (listen UDP) Client A and Client B registers to above Soft-switch. All the above are individual servers (Linux CentOS based) i need help in a (conversion) ser.cfg. Can anyone post a sample ?? below is my ser.cfg (tried using t_relay_to_udp(), rewritehost(), rewritehostport(), forward functions BUT none seems to work the way i want it to) SER 1 == 192.168.10.203 Softswitch == 192.168.10.251 Client A == 192.168.10.221 or .236 (SIPp client-registered to soft-switch successfully) Client B == 192.168.x.x (registered to soft-switch successfully) Thanks shengmai shengmai_goh(a)innomedia.com.sg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei 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) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #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/print.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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(); #if (method == "INVITE") { # if (uri =~ "sip:*@*"){ # log(1, "Forwarding to Vsp12K\n"); # rewritehostport("192.168.10.204:5060"); # t_relay(); # break; # } #}; if (uri =~ "sip:*@*"){ #log(1, "Forwarding to Vsp12K\n"); #rewritehostport("192.168.10.204:5060"); if (uri =~ "sip:*@192.168.10.203"){ rewritehostport("192.168.10.251:5060"); } t_relay_to_udp("192.168.10.251", "5060"); #t_relay(); break; }; # 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); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # ----------------------------------------------------------------- # Pass on stuff going to PSTN to Asterisk # ----------------------------------------------------------------- #if (uri=~"^sip:*@.*") { # rewritehostport ("*192.168.10.204:5060"); # if (!t_relay()) { # # sl_send_reply ("403", "prout"); # sl_reply_error(); # }; # break; #}; rewritehostport("192.168.10.204:5060"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; break; }

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[Serusers] mediaproxy server NEW FEATURE!!!
by Gonzalo J. Sambucaro 24 Feb '08

24 Feb '08

Folks, I developed a new feature in the Media Proxy project of AG-Proyect. With this new feature the Media Proxy Server is now able to manage the problem of the NAT IP change in an established call. It is currently in production and working without any problem. Somebody wants to test it? I would like to know what to do to make this feature available to everybody. Does anybody know? Regards, ------------------------------ Details: ------- Supose that there is an established rtp connection between two endpoints and the media proxy is in the middle doing the relay of the rtp streams, let´s say (MP = mediaproxy) EP A <--------->[NAT with IP1]<-----------> MP <--------------------> EP B EP A sends rtp to MP_IP:MP_PORT passing through the NAT box. EP B sends rtp to MP_IP:MP_PORT without passing through a NAT box. The MP know that the caller = NAT_IP1:NAT_PORT1, and the called = EP_B_IP:EP_B_PORT Now, supose that the NAT box change their PUBLIC IP from IP1 to IP2, so this escenary EP A <------->[NAT with IP1]<---------> MP <-------------> EP B will change to this EP A <------->[NAT with IP2]<---------> MP <--------------> EP B so the MP should detect that change of IPs and continue relaying the rtp streams but now to IP2:PORT2 instead of IP1:PORT1. Well, that was the situation y have experienced. To fix this, I developed this solution, changing rtphandler.py file: 1) When the first rtp packet of a source arrives, save the SSRC field in the MP. - Save the SSRC of the caller. - Save the SSRC of the called. 2) If arrives a rtp packet with unknown source IP but with the same SSRC field of some of the two streams, updates the binding (with the new IP detected) between the caller and the MP or between the called and the MP according to the field SSRC previously saved. Note: SSRC (RFC 3550 RTP), (from the rfc: "The SSRC identifier carried in the RTP header and in various fields of RTCP packets is a random 32-bit number that is required to be globally unique within an RTP session ") -------------------------------- -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo.sambucaro(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

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[Serusers] SIP Express Bundle ready-to-go idea, BUILD FROM CVS on RH ?
by Mike Trest - Personal 24 Feb '08

24 Feb '08

Hi, Followup on prior message: When doing make .... modules I found a similar error that did not involve a system include. The offending line in avpops_db.h is: int db_store_avp( db_key_t *keys, db_val_t *vals, int n, char *table); Both examples below and this example are related to forward declarations with pointers to structures. Regards, ..mike.. make[1]: Entering directory `/usr/local/src/ser-2.0.x-CVS/modules/avpops' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev17-tcp"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.1"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c avpops.c -o avpops.o In file included from avpops_impl.h:41, from avpops_parse.h:38, from avpops.c:50: avpops_db.h:66: error: expected â)â before â*â token make[1]: *** [avpops.o] Error 1 make[1]: Leaving directory `/usr/local/src/ser-2.0.x-CVS/modules/avpops' make: *** [modules] Error 1 ======================================================================== Hi, I am doing a ser build from CVS check-out on a RH based linux with gcc 4.1.1. The build commands are this. make clean make proper make group_include="standard standard-dep stable mysql" When I get to any core or module section code which included futexlock.h header, I get the following: gcc -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev17-tcp"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.1"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c core_cmd.c -o core_cmd.o In file included from mem/../futexlock.h:48, from mem/../lock_ops.h:85, from mem/shm_mem.h:76, from core_cmd.c:32: /usr/include/linux/futex.h:96: error: expected â)â before â*â token /usr/include/linux/futex.h:98: error: expected â)â before â*â token make: *** [core_cmd.o] Error 1 ========================================================================= The offending lines came from the system header <futex.h>. The offending lines are: long do_futex(u32 *uaddr, int op, u32 val, unsigned long timeout, u32 *uaddr2, u32 val2, u32 val3); extern int handle_futex_death(u32 *uaddr, struct task_struct *curr, int pi); By pointing to an alternate version of <futex.h> where these lines are commented out I can compile all the way thru with non-fatal warnings. My questions are: 1) can anyone tell me what are the equivalent lines in the DEB version where this code/compile was most recently developed? 2) has anyone got a fix for this error? ..mike.. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Out-of-office-reply: Serusers Digest, Vol 58, Issue 16
by David.Monahan＠aepona.com 23 Feb '08

23 Feb '08

I will be out of the office until 3 March 2008. I will read your email on my return.

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[OpenSER-Users] SEMS is not working with openser 1 .3
by LetMeKnow 22 Feb '08

22 Feb '08

Hello Users I installed the Sems both version 0.10.0.rc2 and 0.10.0-bundle, Am using OpenSER 1.3 version. When i integrated openser 1.3 and sems both version, is not working i Got this error in openser server *error: mi_datagram: identify_command the request's first line is invalid: no newline after the second error:mi_datagram : mi_diagram_server: command not availble *openser is communicating but sems server send this message to openser server. *debug: cachemsg (AmCtrlInterface.cpp 186) recv-ed: <500 command not available > Error: send_msg (AmServer.cpp:152) AmServer::send_request: ser ansered: 500 c0mmand not available Error: onInvite (EarlyAnnouncement.cpp 93) 500 cound not reply * But with openser.1.2.2 is working with these versions of SEMS... please help me .... -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

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[OpenSER-Users] Probs With TLS Certificate
by Ali Jawad 22 Feb '08

22 Feb '08

Hi All I want to encrypt the communication between the softphone and OpenSER. I have compiled 1.3 with TLS support and I am using the default certs that come with the source code and are located in /usr/local/etc/openser/tls/user My TLS config looks as follows disable_tls = no listen = tls:xx.xx.xx.xx:443 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" I am getting the following error on my softphone ERROR:core:tls_accept: some error in SSL: ERROR:core:tls_print_errstack: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca Any Suggestions ? Thx

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[Serusers] SIP Express Bundle ready-to-go idea, BUILD FROM CVS on RH db_key_t ?
by Mike Trest - Personal 22 Feb '08

22 Feb '08

In the CVS tree, I am trying to track it down original definition of db_key_t Can anyone point me to the place? ..mike..

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[OpenSER-Users] Modify From URI Username
by Sergio Gutierrez 22 Feb '08

22 Feb '08

Hello everybody. Is it possible to modify the From URI username for a particular request? I am trying this, but, openser fails to start: $var(tmp) = "4" + $fU; $fU = $var(tmp); Thanks in advance for your help. Best regards. Sergio Gutierrez

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[Serusers] SIP Express Bundle ready-to-go idea, BUILD FROM CVS on RH ?
by Mike Trest - Personal 22 Feb '08

22 Feb '08

Hi, I am doing a ser build from CVS check-out on a RH based linux with gcc 4.1.1. The build commands are this. make clean make proper make group_include="standard standard-dep stable mysql" When I get to any core or module section code which included futexlock.h header, I get the following: gcc -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev17-tcp"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.1"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c core_cmd.c -o core_cmd.o In file included from mem/../futexlock.h:48, from mem/../lock_ops.h:85, from mem/shm_mem.h:76, from core_cmd.c:32: /usr/include/linux/futex.h:96: error: expected â)â before â*â token /usr/include/linux/futex.h:98: error: expected â)â before â*â token make: *** [core_cmd.o] Error 1 ========================================================================= The offending lines came from the system header <futex.h>. The offending lines are: long do_futex(u32 *uaddr, int op, u32 val, unsigned long timeout, u32 *uaddr2, u32 val2, u32 val3); extern int handle_futex_death(u32 *uaddr, struct task_struct *curr, int pi); By pointing to an alternate version of <futex.h> where these lines are commented out I can compile all the way thru with non-fatal warnings. My questions are: 1) can anyone tell me what are the equivalent lines in the DEB version where this code/compile was most recently developed? 2) has anyone got a fix for this error? ..mike..

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[OpenSER-Users] OpenSER: xcap_auth.c, does it filter other conditions except action/sub-handling, like sphere and va
by KevinKinnan 22 Feb '08

22 Feb '08

My testplatform is openser+openxcap. Only using action/sub-handling as permission rules passed the test. But neither sphere and validity rules works. The file "xcap_auth.c" in presence_xml directory can't give me any clue about sphere and validity. Does openser(presence_xml) support sphere and validity as presence auth rules when fetching xcap doc from OpenXCAP server? Thank you! Kevin, _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

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[Serusers] FW: Intercept SIP call
by roayu 22 Feb '08

22 Feb '08

Resend. _____ From: roayu Sent: Wednesday, February 20, 2008 11:21 AM To: serusers(a)lists.iptel.org Cc: roayu Subject: Intercept SIP call Hi there! Recently, I had setup a SIP environment with few sip phones, SIPphone A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to SIPphone B, but B is not able to answer the call. Can SIPphone C answer A's call? I would like to know whether is SER able to have the intercept sip calls function? If yes, what module or configuration that I need to set? Thanks in advance! Best regards, RoaYu

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[Serusers] Intercept SIP call
by roayu 22 Feb '08

22 Feb '08

Hi there! Recently, I had setup a SIP environment with few sip phones, SIPphone A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to SIPphone B, but B is not able to answer the call. Can SIPphone C answer A's call? I would like to know whether is SER able to have the intercept sip calls function? If yes, what module or configuration that I need to set? Thanks in advance! Best regards, RoaYu

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[Serusers] Still having problems with these ACKs (and BYEs).
by SIP 21 Feb '08

21 Feb '08

While it all works fine for our locally-registered users (and for some of our peers), I'm still having problems from some peers dialing numbers on our service. What happens is that we get an ACK (all our ACKs are set to simply relay), and that ACK bounces around inside our server for a while, then times out without ever replying -- which, after a short period of time, cancels the connection for the peer, since they don't see any ACK replies. We start with a normal ACK sent from the remote peer (198.65.166.131) to our server (22.33.44.55): U 198.65.166.131:5060 -> 22.33.44.55:5060 ACK sip:301@22.33.44.55:5060 SIP/2.0. Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>. Via: SIP/2.0/UDP 198.65.166.131;branch=0. Via: SIP/2.0/UDP 192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh qdm9;rport=2051. Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. From: "User One" <sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t. To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6. Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190. CSeq: 1 ACK. Max-Forwards: 16. Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>. Content-Length: 0. P-hint: rr-enforced. P-NATed-URI: YES (1). P-RTP-Proxy: YES (1). We end up getting things that look like this: U 22.33.44.55:5060 -> 22.33.44.55:5060 ACK sip:301@22.33.44.55:5060 SIP/2.0. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>. Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 22.33.44.55;branch=0. Via: SIP/2.0/UDP 198.65.166.131;branch=0. Via: SIP/2.0/UDP 192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh qdm9;rport=2051. From: "User One" <sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t. To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6. Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190. CSeq: 1 ACK. Max-Forwards: 2. Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>. Content-Length: 0. P-hint: rr-enforced. P-NATed-URI: YES (1). P-RTP-Proxy: YES (1). The top of that message shows that this is a message sent from our server to our server, and it just tacks on a Record-Route and a Via line one after another until it times out. Again, my ACK handler is simply: if(method=="ACK") { route(1); break; } The record route is the very basic: if(method!="REGISTER") { record_route(); }; And route[1] is just: route[1] { if (!t_relay()) { sl_reply_error(); }; } All very basic stuff (and straight from the getting started pages I think), and it mostly all works. Local users have no issues. Some peers are okay. But several of them are NOT, and I've tried everything I can think of and can't get ACKs to behave. Can someone give me a bit of a clue as to what to look at? I don't understand the reason for tacking on another RR param each time (nor do I REALLY understand why the lr=on is tacked on when this is clearly not loose-routed). Some days, I feel like I really have a handle on SER. Other days, I feel like I'm stumbling blindly. I never seem to get to a level of comfort. N.

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[OpenSER-Users] Openser 1.3 crashes with SIGNAL 10 on Solaris
by Sergio Gutierrez 21 Feb '08

21 Feb '08

Hello Everybody. Our deployment of OpenSER 1.3 is crashing repeatedly, announcing the reception of SIGNAL 10 at timer process This is the tail lines of log: Feb 17 11:47:55 [4629] WARNING:core:run_actions: null action list (rec_level=2) Feb 17 11:47:55 [4646] CRITICAL:core:receive_fd: EOF on 16 Feb 17 11:47:55 [4628] WARNING:core:run_actions: null action list (rec_level=2) Feb 17 11:47:55 [4623] INFO:core:handle_sigs: child process 4632 exited by a signal 10 Feb 17 11:47:55 [4623] INFO:core:handle_sigs: core was generated Feb 17 11:47:55 [4623] INFO:core:handle_sigs: terminating due to SIGCHLD Feb 17 11:47:55sig_us]: signal e : signa: r received ce: sig15 rec received and this is the backtrace of the core file: #0 fm_free (qm=0x14d118, p=0x30001516) at mem/f_malloc.c:333 #1 0x000be270 in db_free_columns (_r=0x189d68) at db/db_col.c:39 #2 0xfeda6bc4 in db_mysql_free_dbresult (_r=0x189d68) at res.c:191 #3 0xfeda2db4 in db_mysql_free_result (_h=0x16c658, _r=0x189d68) at dbase.c :249 #4 0xff044234 in get_all_ucontacts (buf=0x189d75, len=152445, flags=64, part_idx=0, part_max=1) at dlist.c:207 #5 0xfee35a20 in pingClients (ticks=60, param=0x0) at functions.h:45 #6 0x00084fa8 in start_timer_processes () at timer.c:275 #7 0x0002ffe8 in main (argc=1169408, argv=0x14c578) at main.c:873 Thanks in advance for any help. Best regards. Sergio Gutierrez.

1 1

[Serusers] SIP Express Bundle ready-to-go idea, MODULE LIST?
by Mike Trest - Personal 21 Feb '08

21 Feb '08

Hi, where can I get a maximum list of 2.0.0-rc1 loadmodule commands in the correct order for a good bundled ser.cfg"? I am really frustrated trying to guess how do load for "lookup_user" and "lookup_doman" to work properly. I have the "rc1" source and the full CVS available. I have the table names correct and the MYSQL is connecting. However, I cannot get a good load module list. ..mike..

2 2

[OpenSER-Users] anybody using group_radius module?
by jh＠tutpro.com 21 Feb '08

21 Feb '08

in the process of merging some of the radius modules, i started to wonder if anyone is really using group_radius module that provides a single function radius_is_user_in(). if i remember correctly, the module was originally written by jan to complement db based group module. i personally don't use group module, because it is more efficient to return caller's or callee's groups as reply items from authentication or radius_does_uri_exist calls. please reply to this if you use group_radius module. if no positive replies arrive, perhaps we can delete the module altogether. -- juha

1 0

[Serusers] consultant needed for small project
by Damon Estep 21 Feb '08

21 Feb '08

Here is a brief description of what I need, please send comments to damon(a)soho-systems.com We are looking for someone to give us a "quick start" with SER. We have a "meshed" asterisk implementation with multiple servers used as registration and feature servers. Everything is 100% asterisk now. We have an asterisk server as a front end DID router that acts as a back to back user agent between the upstream carrier and feature server of the end user. When a call comes in from upstream the front end box does a database lookup (MySQL 5) to determine which feature server the user is expected to be registered on and bridges the call to the feature server. If the user is not registered voicemail or forwarding is handled by the feature server , not front end. No outbound calls are handled by the front end box, and no authentication other than trusted IP address is used by the upstream carriers. There is no need for registration knowledge on the front end box. While this is working extremely well as is, we recognize the scalability advantages that could be obtained with a true SIP proxy in the front. Here is what we would like SER to do; 1. Accept invites from two or more upstream carriers, trusted by IP address and not registered or authenticated in any other way 2. Do a database query on the MySQL 5 database to determine the IP address (or SIP URI) of the registration server the user is expected to be on. 3. Proxy the SIP signaling between the upstream carrier and the appropriate registration server. 4. Record the call record and result in a MySQL database table 5. Remain in the SIP signal path, but stay out of the RTP path. 6. Return a 404 not found when there is no database match for the DID in the database. Initial Questions; Can SER look into a custom database table for routing information on each call? Can SER act as a stateful SIP proxy without the end user registering on it? Other information; Call volume is about 25,000 inbound calls per day. We run 100% of our servers on Linux HA clusters of 2 nodes, we do not need assistance with this portion of the setup as we are very well versed in HA clustering, but it is important to know because during a failover registration state would be lost and therefore static routing based on DB queries is preferred. We have extensive Asterisk experience, no Asterisk work is required. Base OS is CentOS 4.6 SER .9.x with the MySQL module has already been loaded, but not configured Root SSH access can be provided There is NO NAT involved, the upstream, the SER box, and all asterisk feature server sue real IP addresses. Any NAT would be between the UA and the feature server, and the feature server acts as a B2BUA and handles NAT traversal. Calls would never be proxied to the UA directly, only to the feature servers. Please let me know you rate for this type of work, a rough time estimate to do only the configuration of SER (no installing required), your availability, and your past experience with similar implementation. This is for an TSP service thousands of end users, it is a mission critical implementation, so EXPERTS ONLY please. After successful testing plan on an hour or two to walk us through the config and explain the routing logic. We are in the GMT -7 mountain time zone, and English is our only spoken language. We would prefer somebody who could be available during this implementation during our regular business hours of 8AM to 6PM MST. We can pay via paypal if desired. Thanks!

1 0

[OpenSER-Users] Interop dialog/sst - acc module to produce CDR for dialog-expired call
by Schumann Sebastian 21 Feb '08

21 Feb '08

Dear all I am looking for a possibility to somehow assure, that the accounting CDR produced will not be infinite, if a call is interrupted without sending a final BYE. I thought to do it like this: Use SST module to create certain time limit (e.g N) for a session. Use dialog module to cut dialog after certain time expired (e.g. 2N). As long as SST creates active sessions (i.e. the client is alive) the dialog does not expire, if the call is interrupted, the dialog expires worst case with 2N time accounted. The problem is, I cannot generate CDR to stop accounting for a certain call where the dialog expired. Is this possible to do in a way? Thanks for your help. Best regards Sebastian

2 6

[OpenSER-Users] Massive "ERROR: bad header field" lead to crashing of OpenSER 1.2.X ?
by David Loh 21 Feb '08

21 Feb '08

Hi all, Recently my OpenSER log showed a lot of these error, ------------------------------------------------------------------------ Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR:parse_content_length: parse ERROR near char [0][ Feb 20 22:14:47 ser ] Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR:get_hdr_field: bad content_length header Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR: bad header field Feb 20 22:14:47 sip /sbin/openser[3607]: find_first_route: ERROR while parsing headers Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR:parse_content_length: parse ERROR near char [0][ Feb 20 22:14:47 ser ] Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR:get_hdr_field: bad content_length header Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR: bad header field Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR:parse_content_length: parse ERROR near char [0][ Feb 20 22:14:47 ser ] Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR:get_hdr_field: bad content_length header Feb 20 22:14:47 sip /sbin/openser[3607]: ERROR: bad header field Feb 20 22:14:47 sip /sbin/openser[3607]: find_credentials(): ERROR while parsing headers Feb 20 22:14:47 sip /sbin/openser[3607]: pre_auth(): ERROR while looking for credentials Feb 20 22:14:47 sip /sbin/openser[3605]: ERROR:parse_content_length: parse ERROR near char [0][ ------------------------------------------------------------------------ from our past experiences, those error are most likely come from corrupted devices (or device/soft phone that send malformed/non-standard message to our SIP port). If we're continuously having this kind of errors for hours (at a rate of few thousands per hour), eventually OpenSER will not accepting new connection (REGISTER) and may just crash. Wondering if there's anything we can do on OpenSER to prevent this from happening? Thanks, David Loh

2 1

[OpenSER-Users] openxcap question
by Adrian Georgescu 21 Feb '08

21 Feb '08

Klaus It was a design consideration. OpenSER queries for the pres rules document based on app type. If you store in OpenXCAP server multiple pres-rules application documents with different names this had worked but OpenSER will simply pick the first document depending on what mysql query returns first. Finally you have multiple clients that write their own pres-rules but OpenSER care about only one (and you do not know before hand which one as app type is equal). So we setup OpenXCAP server to consider that any moment a single pres-rules document can exist for a give users otherwise OpenSER does not know which document to use. I believe that it should be a single document describing the rules for a given presentity as a is central decision making place, Unless there are better ideas ... Adrian If I PUT a document to http://1.2.3.4/@mydomain/org.openmobilealliance.pres-rules/users/test/pres-… and then PUT another document to http://1.2.3.4/@mydomain/org.openmobilealliance.pres-rules/users/test/pres-… , it will overwrite the first file, thus it does not matter if it is pres-rules or pres-rules2. Is this by design, a wrong configuration, or just a bug?

2 1

[OpenSER-Users] Mediaproxy radius error?
by Max Lock 21 Feb '08

21 Feb '08

Hi Folks, I've got mediaproxy running with openser fine, however I want to enable accounting. I've tested accounting to a database, and that seems fine. however as soon as I try radius accounting, it seems to crash out with the errors pasted below. I'm no coder, but it looks like a global variable called filename is not defined? my mediaproxy.ini radius stanza is as follows and the radius dictionaries are all present: [Radius] secret = secret server = cdrradius authport = 1812 acctport = 1813 dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/dictionary.radius, /usr/local/mediaproxy/dictionary retries = 2 timeout = 3 Any ideas? -Cheers Max Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: Traceback (most recent call last): Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: File "/usr/local/mediaproxy/mediaproxy.py", line 71, in <module> Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: import rtphandler Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: File "/usr/local/mediaproxy/modules/rtphandler.py", line 56, in <module> Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: from accounting import accounting, StopRecord, StopRecordSerializer, QueuedItemProcessingThread Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: File "/usr/local/mediaproxy/modules/accounting.py", line 330, in <module> Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: accounting = RadiusAccountingThread() Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: File "/usr/local/mediaproxy/modules/accounting.py", line 308, in __init__ Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: self.__radius = RadiusAccounting() Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: File "/usr/local/mediaproxy/modules/radius.py", line 50, in __init__ Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: abort("cannot read radius dictionaries: %s" % str(e)) Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: File "/var/lib/python-support/python2.5/pyrad/dictionary.py", line 67, in __str__ Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: if hasattr(self, filename): Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: NameError Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: : Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: global name 'filename' is not defined Feb 20 16:14:19 lhc-voipgate mediaproxy[20721]: exiting. Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: Traceback (most recent call last): Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: File "/usr/local/mediaproxy/proxydispatcher.py", line 65, in <module> Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: import dispatcher Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 39, in <module> Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: from accounting import accounting, StopRecordSerializer, UnserializeError Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: File "/usr/local/mediaproxy/modules/accounting.py", line 330, in <module> Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: accounting = RadiusAccountingThread() Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: File "/usr/local/mediaproxy/modules/accounting.py", line 308, in __init__ Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: self.__radius = RadiusAccounting() Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: File "/usr/local/mediaproxy/modules/radius.py", line 50, in __init__ Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: abort("cannot read radius dictionaries: %s" % str(e)) Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: File "/var/lib/python-support/python2.5/pyrad/dictionary.py", line 67, in __str__ Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: if hasattr(self, filename): Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: NameError Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: : Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: global name 'filename' is not defined Feb 20 16:14:19 lhc-voipgate proxydispatcher[20727]: exiting.

2 1

[Serusers] serctl add users problem
by muhammad akl 21 Feb '08

21 Feb '08

i have setup ser and eveythin went smoothly and configured mysql also to communicate with ser but when i tried to add users using serctl i have the following error : awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed any suggestion i'll be appreciated OS : Solaris 10 Ser_version : 0.9.4

1 0

[OpenSER-Users] Authentication credentials cache
by John Barry 21 Feb '08

21 Feb '08

Hello, I would like to know if OpenSER supports some type of authentication credentials caching, to improve the performance when a non-local authentication service (i.e. RADIUS) is used. If it is not supported, are there any plans to include this functionality in future versions? Bests, JB _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

8 8

[OpenSER-Users] openser+diameter integration--->kindly do reply
by bhargav.b＠renovau.com 21 Feb '08

21 Feb '08

hi every body, i am really in search of somebodies help in finding solution to my problem mentioned below.i hope someone will respond kindfully. i have installed opendiameter and openser successfully and all was going well till yesterday.I need to authenticate a client by enabling communication between openser and diameter. when i try to communicate, the diameter shows me a message "invalid session AVP" . if i run both continuously then diameter client continuously the above message. if somebody have got the solution please insist me.It would be so kind of u Bhargav.b

1 0

[OpenSER-Users] Reg. P-asserted Identity and RFC 3325 support
by Padmaja 20 Feb '08

20 Feb '08

Hi all, does the Openser server version 1.0.1 support P-asserted identity header based on RFC 3325? If yes, is there something that needs to be configured on the server to make this feature to work? Thanks, Padmaja

2 1

[OpenSER-Users] cpl "proxy_recurse"
by Daniel 20 Feb '08

20 Feb '08

Hi all, what does "proxy_recurse" parameter do if it not set to zero ? For example: modparam("cpl-c","proxy_recurse",2) regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] Bug in Displayname Parser?
by Helmut Kuper 20 Feb '08

20 Feb '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to replace Displayname in From field with uac_replace_from("$fn", "sip:1234@exaple.com"); What I found ist, that display name seems to be hold with quotes in $fn instead without qoutes. So after running uac_replace_from I got double quotes in front of and behind display name like this: befor uac_replace-from: From: "abc" <sip:1234@example.com>;tag ... after uac_replace_from: From: ""abc"" <sip:1234@example.com>";tag ... Is this a feature or a Bug? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHu/6V4tZeNddg3dwRAlO5AJ4pSYqO0/fw/DPUGtT2oHMebXw/NQCfSPfD lpUpAmoCuLE5mlEMocBzO5o= =Y3ek -----END PGP SIGNATURE-----

2 1

[OpenSER-Users] [CDRTool] BUG: Wrong price when multiple span
by Adrian Georgescu 20 Feb '08

20 Feb '08

Hi Iñaki, I checked the code and made test calls that span multiple profiles. The calculus is correct. The confusion likely arises because StartTime is printed in each span, while it is the StartTime of the call itself and not of the span. Finally the sum of all spans equals the duration of the call. If you have data that contradicts this I would be happy to look at it again. Regards, Adrian >>>>>>>>>>>>>>>> Hi, CDRTool can calculate the price of a call involving various spans (rates). But it does it wrongly since the start time for each span is set to the start time of the call, so the price is much higher than the correct one. An example: This is a call of more than 3 days of duration. Note that in spans 2, 3 and 4 the "StartTime" is 2008-02-16 15:57:05 in all (the start time of the call), and note also that the "Duration" of spans 2, 3 and 4 are the sum of each span and its previous spans: -- Span: 1 Duration: 28975 s StartTime: 2008-02-16 15:57:05 (Europe/Madrid) ProfileId: 01 Festivo for weekend RateId: Reducida for 0-24h Rate: 0.0100 / 60 s Price: 4.8292 -- Span: 2 Duration: 115375 s <---- WRONG !!! StartTime: 2008-02-16 15:57:05 (Europe/Madrid) <---- WRONG !!! ProfileId: 01 Festivo for weekend RateId: Reducida for 0-24h Rate: 0.0100 / 60 s Price: 19.2292 -- Span: 3 Duration: 144175 s <---- WRONG !!! StartTime: 2008-02-16 15:57:05 (Europe/Madrid) <---- WRONG !!! ProfileId: 01 Laboral for weekday RateId: Reducida for 0-8h Rate: 0.0100 / 60 s Price: 24.0292 -- Span: 4 Duration: 229878 s <---- WRONG !!! StartTime: 2008-02-16 15:57:05 (Europe/Madrid) <---- WRONG !!! ProfileId: 01 Laboral for weekday RateId: Normal for 8-22h Rate: 0.0300 / 60 s Price: 114.9390 -- Total: 163.0765

2 1

[OpenSER-Users] The best way to allow just a few destinations based on prefix?
by Iñaki Baz Castillo 20 Feb '08

20 Feb '08

Hi, which would be the more ellegant way to allow users to call just a few destinations (based on prefix)? Of course I could do it simply by creating a custom table: "allowed_destinations" with columns: [int_prefix] [description] 34 Spain 44 England ... and latter extract the international prefix from the E164 converted RURI and do a SQL query. But I'd like to avoid using avp_db_query() function. I've reading about PDT, LCR and CarrierRoute modules but they seems to be supposed to other aims. The most similar could be PDT but it needs to strip my prefix XD What do you suggest? should I implement my custom method for this? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

7 14

[Serusers] Parallel Hunt groups?
by Jeremy A 20 Feb '08

20 Feb '08

Hello, I have an application that requires an incoming call to ring multiple phones simultaneously until one answers or it times out. After that I need to direct the call to another phone or group. Is this feasible using SER? If so, are there any example configurations I can use as a basis - I have already seen the sequential 'hunt group' example, but I need a parallel ring option. Thanks Jeremy

2 1

Re: [OpenSER-Users] Cpl Outgoing and Incoming
by Iñaki Baz Castillo 20 Feb '08

20 Feb '08

El Wednesday 20 February 2008 10:08:44 Daniel Grotti escribió: > Hi, > but how can I get out of outgoing and enter into incoming script? > for example: > > cpl_run_script("outgoing","force_stateful"); > [...] > lookup("location"); > [...] > cpl_run_script("incoming","force_stateful"); Yes, now I understand your main problem. IMHO the only solution is to send outbound the INVITE (so it comes again to OpenSer by DNS resolution). Something as: if ($si != "IP_OPENSER") { cpl_run_script("outgoing","force_stateful"); t_relay(); # Send out the INVITE. } lookup("location"); cpl_run_script("incoming","force_stateful"); A calls B, the INVITE arrives to the proxy from an IP different than the proxy IP, so the outgoing CPL is runned and the CPL by itself will send out the INVITE. The INVITE will come back to the proxy since B is a local user, but in this case the $si is the proxy IP so the lookup and incoming CPL will be runned. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[OpenSER-Users] Cpl Outgoing and Incoming
by Daniel 20 Feb '08

20 Feb '08

Hi all, I'd like to use either cpl Outgoing (of the caller) and cpl incoming script (of the callee). I mean, if "A" wanted to call "B", and "A" and "B" have CPL script saved on proxy, then the message sent by "A" have to go through CPL-OUTGOING of "A" and then through CPL-INCOMING of "B". How can I get this behaviuor in openser.cfg ? Thanks, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

3 3

[OpenSER-Users] Is there OPTIONS keepalive in TCP?
by Iñaki Baz Castillo 20 Feb '08

20 Feb '08

Hi, I register a X-Lite (TCP client) and a Twinkle (UDP client) behind NAT. I dissable STUN, ICE, keepalive, "discover external address", etc... in both. They send a REGISTER to my OpenSer with public IP so I enable OPTIONS pinging ifor both (I confirm that "location" table has the same cflags for the you entries and so, all is correct. But I just see a periodical SIP OPTIONS by UDP for Twinkle. Are they exist in case of TCP? -- Iñaki Baz Castillo

5 17

[OpenSER-Users] Out of pkg memory
by Sergio Gutierrez 20 Feb '08

20 Feb '08

Hello everybody. Recently our deployment of OpenSER 1.3 on Sun Solaris (SPARC) failed with the following error: eb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16935]: [ID 589382 daemon.error] ERROR:core:build_res_buf_from_sip_req: parse_headers failed Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16935]: [ID 765423 daemon.error] ERROR:sl:sl_send_reply_helper: response building failed Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16939]: [ID 515455 daemon.error] ERROR:core:get_hdr_field: out of pkg memory Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16939]: [ID 176737 daemon.error] ERROR:maxfwd:is_maxfwd_present: parsing MAX_FORWARD header failed! Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16939]: [ID 515455 daemon.error] ERROR:core:get_hdr_field: out of pkg memory Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16939]: [ID 517476 daemon.error] ERROR:rr:find_first_route: failed to parse headers Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16939]: [ID 515455 daemon.error] ERROR:core:get_hdr_field: out of pkg memory Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16939]: [ID 697802 daemon.error] ERROR:auth:get_realm: failed to parse TO headers Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16939]: [ID 312964 daemon.error] ERROR:auth:pre_auth: failed to extract realm Feb 19 15:33:33 shannon /opt/local/toip/sbin/openser[16939]: [ID 515455 daemon.error] ERROR:core:get_hdr_field: out of pkg memory What could be the cause of this error? What to check? Thanks in advance for your attention. Best regards. Sergio Gutiérrez

2 1

[Serusers] Strict routing problem
by Rosa De Santis 19 Feb '08

19 Feb '08

Hi List. Please, I need some help with strict routing problem. Incoming calls are canceled at 30 secs from an origination provider. The problem seems to be a bad ACK sent from the origination, because it performs strict routing instead of loose routing. The ACK goes from originator to my SER server, then it must go to a Nat Traversal server and finally to the UA client, but instead of that the ACK tries to go directly from SER to the UA in its local address 192.168.1.109, skipping the Nat Traversal server. The UA sends a BYE to the call because it never receives this ACK. I use another origination provider in the same scenario without problems. This other origination uses loose routing, and ACK messages go in the right path: origination -> SER -> Nat traversal -> UA. How can I resolve this strict routing problem? Thanks in advance, Rosa. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[OpenSER-Users] IRC meeting minutes for today
by Henning Westerholt 19 Feb '08

19 Feb '08

Hi all, the (slightly edited) minutes from the todays IRC meeting are available at: http://www.openser.org/dokuwiki/doku.php/development:irc-meeting-minutes-19… Cheers, Henning

1 0

[OpenSER-Users] openxcap question
by Klaus Darilion 19 Feb '08

19 Feb '08

Hi! If I PUT a document to http://1.2.3.4/@mydomain/org.openmobilealliance.pres-rules/users/test/pres-… and then PUT another document to http://1.2.3.4/@mydomain/org.openmobilealliance.pres-rules/users/test/pres-…, it will overwrite the first file, thus it does not matter if it is pres-rules or pres-rules2. Is this by design, a wrong configuration, or just a bug? thanks klaus

1 0

[OpenSER-Users] Perl module and moduleFunction()
by kionez 19 Feb '08

19 Feb '08

Hi all, i'm trying to write some custom perl module to extend functionalities of openSER. I can't notify UAC with classic sl_send_reply, in my debug log i found: ERROR:core:moduleFunc: Module function 'sl_send_reply' is unsafe. Call is refused. ERROR:core:XS_OpenSER__Message_moduleFunction: calling module function 'sl_send_reply' failed. Missing loadmodule? in my perl script i put a simple: $m->moduleFunction("sl_send_reply", "200", "OK"); as i read in module's example i found in samples/function.pl file. there's some way to send replyes in perl scripts? thanks in advance. k.

2 1

Re: [OpenSER-Users] [OpenSER-Devel] next irc meeting
by Bogdan-Andrei Iancu 19 Feb '08

19 Feb '08

Hi there, I just uploaded on the wiki my wish list for tomorrow ;). Henning, do we count on you as moderator ( as usually ;) ) ? Regards, Bogdan Henning Westerholt wrote: > On Tuesday 12 February 2008, Daniel-Constantin Mierla wrote: > >> Hello, >> >> I think is time for a new irc meeting to synchronize on the work for 1.4 >> and discuss 1.3.1 about release. I propose Friday, Feb 15, UTC 13:00. >> >> I created a wiki page to gather the discussion items there and have the >> agenda ready for that date. I added what I would like to touch. Please >> add yours and complete/comment on mine or others. >> > > Hi Daniel, > > good idea! I've already added some of my points to the list. This friday is > not a good date for me, as i'm on vacation this day. Perhaps we can go for > the Tuesday, Feb 19, UTC 13:00? > > Cheers, > > Henning > > _______________________________________________ > Devel mailing list > Devel(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/devel > >

3 3

[OpenSER-Users] Presence problems
by Leonard Broman 19 Feb '08

19 Feb '08

Hi! Since my last question on this list I have set up the OpenXCAP and put the necessery documents in the database. I however still have the same problem. On subscribe requests the server responds 481 Subscription does not exist and logs: Jan 21 16:04:24 m2lin3 ./openser[5109]: ERROR:presence:get_stored_info: record not found in hash_table Jan 21 16:04:24 m2lin3 ./openser[5109]: ERROR:presence:handle_subscribe: getting stored info Anyone that can tell me why it's behaving like this? Leo

2 1

[OpenSER-Users] Integration of xcap server with openser presence module
by Vincent Nogues 19 Feb '08

19 Feb '08

Dear, I would like to test the open xcap service. So i have installed the openser presence module 1.2 and also the openxcap package but it does not work properly. I have a few questions : Where can I find an up to date installation documentation ? How to make the link between the openser module and the openxcap package ? I have the following errors, someone can help me ? 0(24573) PRESENCE:get_xcap_tree:The query in table xcap for [username]=bob , domain=open-ims.test returned no result 0(24573) PRESENCE: get_subs_dialog:The query for subscribtion for [user]= vno,[domain]= open-ims.test for [event]= presence.winfo returned no result 0(24573) PRESENCE:query_db_notify: Could not get subs_dialog from database 0(24573) PRESENCE:update_subscription:Could not send notify for presence.winfo 0(24573) PRESENCE:get_xcap_tree:The query in table xcap for [username]=vno , domain=open-ims.test returned no result 1(24574) PRESENCE:get_xcap_tree:The query in table xcap for [username]=alice , domain=open-ims.test returned no result 1(24574) PRESENCE:get_xcap_tree:The query in table xcap for [username]=vno , domain=open-ims.test returned no result 1(24574) PRESENCE:get_xcap_tree:The query in table xcap for [username]=bob , domain= open-ims.test returned no result Thanks, Vincent

2 1

[OpenSER-Users] openser 1.3.0 and presence module
by Sigrid Thijs 19 Feb '08

19 Feb '08

Hi, we're testing openser 1.3.0 on a linux server with presence module. I'm using the provided configuration file with the following modifications: - loaded the modules mysql, presence, presence_xml - set the following parameters for these modules: modparam("presence|presence_xml", "db_url", "mysql://openser:[secret]@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:[ip]:5060") modparam("presence_xml","integrated_xcap_server", 1) - uncommented the code to handle PUBLISH/SUBSCRIBE requests to the presence module I also configured the mi_datagram module, to receive mi commands. I discovered the following issues. In the main route block, the second if test checks if the request has a to-tag. If the request does not contain a Route header a 404 Not Here response is sent to the request. When a client subscribes for presence, a 202 OK response is received from the presence module, without Record-Route(s). This means future in-dialog requests won't contain the Route header. It also means that for example an un-SUBSCRIBE request is rejected with a 404 Not Here. I worked around this by removing the has_totag () check and not sending a 404 Not Here response when loose_route () returns false. Furthermore I noticed that sometimes, the presence NOTIFY requests do not contain a body. I checked the database, and there was data in the body column in the presentity table for the presence resource. The value in the status column of the watchers/active_watchers table is 1. When I restart openser, the presence descriptions are added to the NOTIFY requests again (the entry in the database did not change). I have to say that we insert some presence states directly in the database via another process. But this is nothing more than an INSERT or UPDATE statement on the presentity table. Do you have any idea why a NOTIFY without body is sent? Do i need to configure something extra? And why does it seem to work after openser has restarted? kind regards, Sigrid

4 6

[OpenSER-Users] Regarding RLS Presence Server
by Noble Antony T 19 Feb '08

19 Feb '08

Hi, I have configured Openser as an RLS server. Whenever I send a Subscribe for a resourcelist [ with Reqest-URI as resource list ], I am getting NOTIFY message with empty message body. { I was simply giving some junk list name in request URI} now what I am bothered is the configuration of resource list. how will I form a resource list of a number of subscribers..? Is it possible to use it without an XCAP Server..? *"global lists" *and *"users lists".* How do I form a *"users list"*..? with out using an XCAP..? Is there an RFC to specify the local list ( aka "*users list*" where the sender specify whom all he would like to subscribe for Presence Information ) Thanks, -- Noble Antony T.

2 1

Re: [OpenSER-Users] Radius Acct Syntax Error?
by Drix Tabligan 18 Feb '08

18 Feb '08

Thanks Everyone. My mistake, didn't see the ; at the end. VIM colored it dark blue on a black background... Appreciate you help guys... Drix > > Message: 6 > Date: Mon, 18 Feb 2008 12:43:39 +0100 > From: Henning Westerholt <henning.westerholt(a)1und1.de> > Subject: Re: [OpenSER-Users] Radius Acct Syntax Error? > To: users(a)lists.openser.org > Cc: Drix Tabligan <drix(a)tabligan.net> > Message-ID: <200802181243.39271.henning.westerholt(a)1und1.de> > Content-Type: text/plain; charset="ansi_x3.4-1968" > > On Friday 15 February 2008, Drix Tabligan wrote: >> Hi, >> >> A quick help from anyone please? >> >> I've compiled Openser to support radius accounting and did the >> necessary installations already. Freeradius is running and can do >> accounting and auth already when tested. >> >> I'm having a problem starting Openser 1.2. The output shows... >> [..] >> 0(11900) parse error (224,21-22): syntax error >> 0(11900) parse error (224,21-22): >> ERROR: bad config file (2 errors) > > Hi Drix, > > look at line 224, the character at position 21 produces this error. > The error > messages is difficult to understand, this was improved 1.3. > > Cheers, > > Henning > > > > ------------------------------ >

1 0

[OpenSER-Users] TM module is not 3261 compliant (matching ACK to 2xx)
by Taisto Qvist 18 Feb '08

18 Feb '08

Hi again, I hope I am not being to pushy/obnoxious now, but I was really hoping I could get an anwer from the openser-guru's concerning my issues with the transaction matching code I mentioned in an earlier mail. I and my co-workers feel that the RFC is quite clear on the subject of matching ACK to 2xx, so I would really like to hear what the objections are, since openser does not behave this way. What I was trying to do, was simply to stop the ACKs for 3++ in my script by using "t_check_trans()" and if the ACK did match the transaction, the script would end processing. But I suppose I'm supposed to do this some other way?? Anyway, I hope to hear from someone, indicating what your views on this txn-matching issue is. Regards Taisto Taisto Qvist wrote: > I seems like we dont agree on the interpretation of the txn handling > for invite. > Sure, there is a timer for handling retransmitted responses, but thats > in earlier states than Terminated. > Also, there is nothing that says that the inner workings of your transactions > must exactly behave as the statemachine diagrams in the rfc as long as the > external behavior stays the same, but still I think that the ACK for 2xx > should not match the INVITE txn and I believe the RFC is faily clear on this. > (It probably boils down to how rfc-isch you want to be.) > > > [Bogdan] > >The end-to-end ACK establish a separate transaction (RFC 3261) and these > >ACKs are not match as part of the INVITE transactions, but correlated > >with them. > > [TQ] > Where does it say that ACKs establish a separate transaction? > There is no defined ACK transaction, only INVITE/nonInvite, server and client. > The ACK is either a part of the INVITE txn, or its a separate request, but I would > never call it a standalone transaction. > The real purpose of txn's (in my view), is to enable/simplify forking, and to > handle retransmissions. Retransmissions of ACK and 2xx, are done by the UAC/UAS, > so there is no need for a ack-txn. > > > [Bogdan] > > but even describe the wait timer. So, there is no contradiction. > > [TQ] > I'd say there is. Where does it describe that this wait-timer should be > used for all responses, or for matching *all* ACKs? > > My main reasons for saying that ACKs for 2xx should not be matched, > come from the following texts. (17.1.1.2, and 17.2.1) > > 17.1.1.2 (client invite txn) > The client transaction MUST be destroyed the instant it enters the > "Terminated" state. This is actually necessary to guarantee correct > operation. The reason is that 2xx responses to an INVITE are treated > differently; each one is forwarded by proxies, and the ACK handling > in a UAC is different. Thus, each 2xx needs to be passed to a proxy > core (so that it can be forwarded) and to a UAC core (so it can be > acknowledged). No transaction layer processing takes place. > Whenever a response is received by the transport, if the transport > layer finds no matching client transaction (using the rules of > Section 17.1.3), the response is passed directly to the core. Since > the matching client transaction is destroyed by the first 2xx, > subsequent 2xx will find no match and therefore be passed to the > core. > > [TQ] > It even says "actually necessary to guarantee...." > Since the txn's are there to (among other things) absorb retransmissions, > the receiving the 2xx MUST destroy the txn, so that when the next retransmission > of 2xx is received, it is not consumed by the txn layer. > This is what your txn-layer does for retransmitted 3++ right? > Any additional 3++ received on a txn while waiting for Timer D to expire, > will just be consumed, and the proxy core will never have to process it. > Or? > > 17.2.1. (server invite txn) > The purpose of the "Confirmed" state is to absorb any additional ACK > messages that arrive, triggered from retransmissions of the final > response. > ..snip... > Once the transaction is in the "Terminated" state, it MUST be destroyed > immediately. > > [TQ] > The Confirmed state is there to absorb retransmissions, not the terminated. > When receiving 2xx you go directly to terminated, bypassing, confirmed. > > Also, in 18.2.1 the text even explicitly says(sure, not "MUST" but..): > > .. Note that when a UAS core sends a 2xx response to INVITE, > the server transaction is destroyed. This means that when the ACK arrives, > there will be no matching server transaction, and based on this rule, > the ACK is passed to the UAS core, where it is processed. > > > <snip old stuff> > > What I would dream of, is simply have some function-parameters to the > t_check_trans(<param>), or even better in my little mind, > a "modparam("tm", "ack_2xx_matching", 1)". > > I am looking forward to hearing your reply, and in the mean time, > thanks to all of you developers for an extraordinary software :-) > > Regards > Taisto Qvist > > > Hi guys, > > > > Just to bring some clarifications on the TM module. > > > > once a transaction is completed (negative or 2xx reply), it is put on > > wait (wait timer - see RFC3261) for catching any potential > > retransmissions of replies. > > So, the transaction is completed, but not removed from memory - RFC does > > not say that you need to trash immediately all the transaction > > information, but even describe the wait timer. So, there is no > > contradiction. > > > > The ACK (for 2xx)catching is done based on the completed INVITE > > transaction (from wait timer) - nothing else. > > > > > > Inaki, could you please detail what you mean by: > > > > <quote> > > In my opinion OpenSer does a special treatment for ACK in tm mode, even if > > they are for failed transaction (hop-by-hop ACK's) or succesfull INVITE > > (end-to-end ACK's). > > > > </quote> > > > > Maybe I can explain you more if I understand you question.... > > > > Regards, > > Bogdan >

2 1

[OpenSER-Users] openser is adding itself as a route
by Lazer Ramm 18 Feb '08

18 Feb '08

Hi, I am running openser 1.2.3 on ip 10.10.10.10 openser is adding itself as a route and the next hop then sends the invite back to it. Is there a way to strip the route when sending out the invite? See >> below<< INVITE sip:192.168.10.1:5060;transport=UDP SIP/2.0..Record-Route: <sip:10.10.10.10;lr=on>.. --- >>>> this is the problem >>> Route: <sip:16464894489@10.10.10.10;user=phone>.. <<<< <<<<<< Via: SIP/2.0/UDP 10.10.10.10;branch=z9hG4bK9ffa.8103bc92.0..Via: SIP/2.0/UDP 172.16.100.17:506 0;branch=z9hG4bKckr04nq099hc7oit3g6bl0s;rport=5060..From: <sip:706@10.10.10.10>;tag=k1404nvk7hhc61333g66..To: <sip:164 64894489(a)10.10.10.10;user=phone>..Contact: <sip:706@172.16.100.17;transport=UDP>..Supported: 100rel,sec-agree..CSeq: 1 153 INVITE..Call-ID: ZII2nVPhoIfc11McnXIVhs0adjyRuT..Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK..Expires: 1 20..Privacy: none..User-Agent: Nokia RM-208 1.0633.18.01..P-Preferred-Identity: sip:706@10.10.10.10..Max-Forwards: 70. .Content-Type: application/sdp..Accept: application/sdp..Content-Length: 435....v=0..o=Nokia-SIPUA 63371528233679000 633 71528233679000 IN IP4 172.16.100.17..s=-..c=IN IP4 172.16.100.17..t=0 0..m=audio 16384 RTP/AVP 96 0 8 97 18 98 13..a=pti me:20..a=maxptime:200..a=fmtp:96 mode-change-neighbor=1..a=fmtp:18 annexb=no..a=fmtp:98 0-15..a=rtpmap:96 AMR/8000/1..a= rtpmap:0 PCMU/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:97 iLBC/8000/1..a=rtpmap:18 G729/8000/1..a=rtpmap:98 telephone-ev ent/8000/1..a=rtpmap:13 CN/8000/1.. #

3 2

Re: [OpenSER-Users] SIP Trunk doesn't response to 200OK with an ACK
by Eric Phetteplace 18 Feb '08

18 Feb '08

Sorry about that: Network Topology. 8.21.8.24 is the PSTN gateway (WAN IP) 192.168.2.5 UAS (simple IP phone, "PC to PC" no registrar, etc) 192.168.2.1 NAT Router with 192.168.2.10 in DMZ WAN Address is 2.6.7.9 in my previous message. 192.168.2.10 OpenSER and RTPProxy.

3 2

Re: [OpenSER-Users] OpenXcap + Presence
by Andreas M. 18 Feb '08

18 Feb '08

Hello, still playing arround with presence and xcap, another thing i would like to know is following query that i can see in my sql logs: username,domain,doc_type,etag,doc_uri,port from xcap where source=1 Q: What does the "source" value stand for, because currently all data records have a value of "0" Also trying currently with "inetgrated xcap_server="0" " but with "xcap_client" module, also with no success... Presence status keeps always pending.... Any hints ? r, Andreas M. Andreas M. schrieb: > Hello, > i installed OSER 1.3-SVN and openxcap-0.9.9 (does not work with OSER 1.2 :) ) > > XCAP Server is running, also openser, but i can´t see any presence status, clients sends publish but > no notifys are sent out. > > i´m using latest eyebeam clients, i can also see that privacy rules are updated in the xcap > database, but subscribtion status is always pending. > > i can see that xcap doc is loaded through a query, but status is pending. > Had a long time presence running mit "force_active = 1", but now i want to introduce also xcap > functionality. > > Where can i start to debug ? > > r, > Andreas M. > > presence config: > ## presence module > > modparam("presence", "db_url","mysql://xx:xx@127.0.0.1/presence") > 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", 'mango') > modparam("presence", "expires_offset", 10) > modparam("presence", "max_expires", 3600) > modparam("presence", "server_address", "sip:10.0.0.1:5065") > modparam("presence", "fallback2db", 1) > > modparam("presence_xml", "db_url","mysql://xxx:xxx@127.0.0.1/presence") > modparam("presence_xml", "xcap_table", "xcap") > modparam("presence_xml", "force_active", 0) > modparam("presence_xml", "pidf_manipulation", 0) > modparam("presence_xml", "integrated_xcap_server", 1) > > also i can see following query in my sql log, that does not work (field does not exists) > > select id from watchers where inserted_time<1202992297 AND subs_status='pending' > > source: > CREATE TABLE `watchers` ( > `id` int(10) unsigned NOT NULL auto_increment, > `presentity_uri` varchar(128) NOT NULL, > `watcher_username` varchar(64) NOT NULL, > `watcher_domain` varchar(64) NOT NULL, > `event` varchar(64) NOT NULL default 'presence', > `status` int(11) NOT NULL, > `reason` varchar(64) default NULL, > `inserted_time` int(11) NOT NULL, > PRIMARY KEY (`id`), > UNIQUE KEY `watcher_idx` (`presentity_uri`,`watcher_username`,`watcher_domain`,`event`) > ) ENGINE=MyISAM AUTO_INCREMENT=10 DEFAULT CHARSET=latin1; > > -- g, Andreas M.

1 0

[OpenSER-Users] copy/paste bug in module documentation
by Klaus Darilion 18 Feb '08

18 Feb '08

http://openser.org/docs/modules/1.3.x/xcap_client.html The module documentation has the wrong header (Presence XML) klaus

2 1

[OpenSER-Users] Radius Acct Syntax Error?
by Drix Tabligan 18 Feb '08

18 Feb '08

Hi, A quick help from anyone please? I've compiled Openser to support radius accounting and did the necessary installations already. Freeradius is running and can do accounting and auth already when tested. I'm having a problem starting Openser 1.2. The output shows... --------------- 0(11900) set_mod_param_regex: acc matches module acc 0(11900) set_mod_param_regex: found <service_type> in module acc [/ usr/local/lib64/openser/modules/] 0(11900) set_mod_param_regex: acc matches module acc 0(11900) set_mod_param_regex: found <radius_config> in module acc [/ usr/local/lib64/openser/modules/] 0(11900) parse error (224,21-22): syntax error 0(11900) parse error (224,21-22): ERROR: bad config file (2 errors) --------------- If I comment out the modparam("acc", "radius_config", "/usr/local/etc/ radiusclient/radiusclient.conf") below, Openser runs. A snip of the openser config... --------------- modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "service_type", 15) modparam("acc", "radius_config", "/usr/local/etc/radiusclient/ radiusclient.conf"); modparam("acc", "radius_flag", 0) modparam("acc", "radius_missed_flag",0) modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i: 902)") modparam("acc", "radius_extra", "Sip-RPid=$avp(s:rpid); \ Sip-Src-IP=$si; \ Sip-Src-Port=$sp; \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$avp(s:rtp_statistics); \ From-Header=$hdr(from); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip)") -------------- Which means that radius accounting was compiled in as the other parameters doesn't spew the error. The radiusclient.conf and dictionary files are essentially untouched except for the necessary changes on the shared secret and ip addresses for the radius config. Openser, Freeradius and Radiusclient are running on the same box under FC7 64bit. Thank you in advance for your help... Drix

2 1

Re: [OpenSER-Users] Failover using NAPTR/SRV
by Douglas Garstang 18 Feb '08

18 Feb '08

Ok, so rfc3263 defines what the user agent should do when a 503 reply, or no reply is received. What about the rest of the possible SIP replies? We use several different ITSP's. I just checked the logs for our last 50,000 calls from all of them and there was a total of 19 different replies received. They are: 200,400,401,403,404,408,410,480,484,486,487,488,500,501,502,503,504,603 Some of these can occur under normal conditions, 486 is busy for example, or 487 is generated by a CANCEL. What about the rest? Most of the rest can be classified as error conditions. Are you saying that if another error condition besides 503 is received that you don't do failover? Why? Douglas. ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: Douglas Garstang <dougmig33(a)yahoo.com>; users openser.org <users(a)openser.org> Sent: Friday, February 15, 2008 5:56:45 AM Subject: Re: [OpenSER-Users] Failover using NAPTR/SRV Of course it is.....by IETF - see RFC3263. 4.3 Details of RFC 2782 Process Regards, Bogdan Douglas Garstang wrote: > Well that's weird. What do you mean by 'fail'? I thought dns_blacklist > was used when a 503 response was received? What if the connection > times out? What if another negative reply is received? Is this stuff > documented anywhere? > > ----- Original Message ---- > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > To: Tobias Lindgren <tobias.lindgren(a)ip-only.se> > Cc: users(a)openser.org > Sent: Thursday, February 14, 2008 3:25:33 AM > Subject: Re: [OpenSER-Users] Failover using NAPTR/SRV > > Hi Tobias, > > if you have "dns_backlist=yes" in your config, if one of the destination > server fails (according to SIP definition), it's IP will be added to a > temporary blacklist (for 4 minutes) and not used. So, openser should do > dns-based failover and use the next entry provided by NAPTR/SRV/A lookup. > > Regards, > Bogdan > > Tobias Lindgren wrote: > > Hi all, > > > > I've been trying to find this information but I cannot find any exact > > specifications on how it really works. > > > > >From what I know using NAPTR/SRV records with OpenSER will allow > it to > > find and use servers behind those DNS-records. This works just fine. > > > > However, what I'm not sure about is what actually will happen in > OpenSER > > when one of two servers in this scenario would fail. > > > > For example, I have two servers as SRV where one is primary and one is > > secondary for SIP/UDP. What will happend in OpenSER when the primary > > server is down? Will OpenSER continue to send all request first towards > > that server or will it learn that one server is down and always send > > requests to the second server for a period of time and try the primary > > one just occassionally? > > > > Please direct me to any page where this is explained in detail, if such > > page exists. > > > > Br, > > /Tobias > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > ------------------------------------------------------------------------ > Never miss a thing. Make Yahoo your homepage. > <http://us.rd.yahoo.com/evt=51438/*http://www.yahoo.com/r/hs> ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[OpenSER-Users] OpenSER VS. SER
by Ali Jawad 18 Feb '08

18 Feb '08

Hi All I would like to know if anyone can state the most important differences between openSER and SER in terms of features. I am not very interested in CPU and RAM consumption I would like to know if there are any differences between the two servers in terms of functionality. Thanks

4 4

[OpenSER-Users] Survey for new mailing list
by Bogdan-Andrei Iancu 18 Feb '08

18 Feb '08

Hi all, I the last year there was a good progress on the project policy regarding releases (major and minor). Still, from my experience as booth developer and user of openser, I feel the need for way to promptly and focused inform the users of openser about the fixes, especially critical fixes, that require immediate update. Many project have lists where you can subscribe in order to get notification for critical bug or security fixes. Currently, for openser, if you are not subscribed on devel list and if you are not "good" enough in "decrypting" the commit logs, you may miss important fixes you may want to update. I think is our duty as project to inform the users about the discovered flows in the stable versions. My suggestion is to create a new mailing list where, who is interested, will receive notifications when something critical was fixed in the stable versions of openser. Maybe a new list (like "alerts(a)lists.openser.org") where a developer, after making the fix, can issue (if considered important) an update alert for the interested people : Severity: Affected services/modules: Description: Rev number where the fix is available: How to update: ? I come to this as right now I'm working at two major fixes in the stable versions affecting two important serviced - user location and accounting. And I bet a lot of people will be interested in this in order to update their installations. Please let me know your thoughts on this. Best regards, Bogdan

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