sr-users October 2005

sr-users@lists.kamailio.org

214 participants
457 discussions

by Helge Waastad

Forgot CC on this one..... -----Original Message----- From: Helge Waastad Sent: 26. oktober 2005 13:56 To: 'Zen Kato' Subject: RE: [Users] SIP clients and DNS SRV Hi, You know, even if DNS SRV works fine, you will have problems with NAT'ed clients behind firewalls. Ther reason is that the session is opened between the server handeling the registration and the UAC. If this on crashes, the second one will never be able to send INVITE to the UAC since there is no open session through the firewall. Yes, you will see service unavaliability up to the registration interval, depending on the UAC functionality. The only solution for this is a (outbound) proxy cluster using Linux-HA (or vrrp) and a IP address take-over. (Or a Border Controller, but these are not for free..) Cisco 7960 works fine (also snom) I've seen that Sipura in 3.1.5 release also supports dns srv for redundancy/failover, but I have'nt tried it. -----Original Message----- From: Zen Kato [mailto:zenkato@pis.bekkoame.ne.jp] Sent: 26. oktober 2005 13:45 To: Helge Waastad Cc: klaus.mailinglists(a)pernau.at; users(a)openser.org Subject: Re: [Users] SIP clients and DNS SRV Hi, I posted on [serusers] regarding DNS SRV,t_replicate,backup server and GS phones. My test resut of BT101(1.0.7.11-newest) and GXP2000(1.0.1.9) was no good. My question is eventhough we set up DNS SRV on server side, if sip clients does not suppot DNS SRV, do we have a risk of (5+5)minutes out of service time when we set up 5 minutes REGISTER intervals? If UA have multiple REGISTERs outgoing lines such as GXP2000, we could set up Line1 for sip1&sip2(DNS SRV) Line2 for sip3 server. In this case if line1 does not work, the user can use Line2. Xlite can register 3 sip servers, but only default sip proxy can use as outgoing line, so the Xlite users have (5+5)minutes out of service risk. Does Cisco 7960 work perfectly on DNS SRV? Regards, Zen Helge Waastad wrote : > Hi, > I've started to add some more information to voip-info. > I'm still waiting for response from Grandstream and others on their > support (or future support) for DNS SRV. > > br hw > > On Wed, 2005-10-19 at 17:32 +0200, Klaus Darilion wrote: > > Hi Helge! > > > > According to my tests, also SNOM can do SRV based failover. > > > > I've published my test results at: > > http://www.voip-info.org/wiki/view/SRV+implementations > > > > Please add your test results to this. > > > > regards > > klaus > > > > Helge Waastad wrote: > > > Hi, > > > this is not a technical question regarding OpenSER, but more of > > > req for information. > > > > > > I'm using Cisco phones and gateways and have implemented DNS SRV > > > redundancy. > > > However, I have never seen a SIP phone that really works with DNS > > > SRV redundancy, except from Cisco. (which actually works as a > > > charm) > > > > > > Does anyone know other phones that can use dns srv for redundancy? > > > > > > At least Cisco tries the next DNS SRV entry if it receives a port > > > unreachable from the highest priority proxy... > > > > > > > > > -- > mvh/best regards > Helge Waastad > System Engineer > Smartnet > (+47)67830017 > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

18 years, 11 months

Re: [Serusers] AVPOPS syntax help

by Samuel Osorio Calvo

I haven't played with AVP much so I might be wrong, but I think you should replace the $ from the username after the alias: avp_db_load($alias/$username,i:alias")---> avp_db_load($alias/username,i:alias") ^ And about the pushto....I have no idea if it accepts two headers, so somebody with more experience will tell you if it is possible to push the alias into the ruri and to in the same avp_pushto. Just a guess, try to do it in two different commands. Hope it helps, Samuel. P.D. Espero q algú et pugui ajudar...sort!! Unclassified >>> Pol <kuroki(a)gmail.com> 10/19/05 09:27PM >>> Hi SerUsers! I'm completly crazy tryin' to do the following: I just want to load from DB user alias and rewrite it into "$to $ruri", because these calls shoud go directly to asterisk voicemail and they are numerical not alphanumeric. i tried to do the following: if(avp_db_load($alias/$username,i:alias") avp_pushto("$ruri/$to","alias") just to load from alias table username into an integer called alias and then pushed to new $ruri. I'm new in ser and avps module it's really usefull but a little bit "unreadable", plz this should work for tomorrow :$ any help plz!? be as much especific as can, because i don't know if i have done the proper modparams. these are: modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") #modparam("avpops", "use_domain", "1") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") THX On 10/13/05, Iqbal <iqbal(a)gigo.co.uk> wrote: > > > Hi > > I think you have the source syntax written below in the avp_db_load > statements > > so the source will have $from or $to or $ruri optionally followed by the > words "username" or domain > > eg > > avp_db_load($from/username) of avp_db_load($to/domain) ---note these are > not completed syntax > > OR instead of the sip_uri which is $from,$to,$ruri you can have avp_alias > > which is exactly what it says, its an alias, > > Str_value this is the number directly related to the predefined alias in > the line above > > The name can be a string S: > interger i: > > or you can leave out the S: and just use the name > > or of course an alias > > I know thats probably not clear...in fact I just read it myself, the best > thing with avpops , is to look at the db, and create a few, and then see > how to pop them out and play with 'em > > Iqbal > > > On 10/12/2005, "Lenir" <lenirsantiago(a)yahoo.com> wrote: > > >Hello list, > > > >I'm reading the documentation for AVPOPS, and Im a little confused in this > >section: > > > >---------------------------------------------------------------------------- > >----- > >#avp_db_load(source,name) > >#Loads from DB into memory the AVPs corresponding to the given source. > > > >Meaning of the parameters is as follows: > > > > > >source - what info is used for identifying the AVPs. Parameter syntax: > > > > > >source = (sip_uri)['/'('username'|'domain')]) | (avp_alias) | str_value > > > >sip_uri = '$from' | '$to' | '$ruri' > > > > > >name - which AVPs will be loaded from DB into memory. Parameter syntax is: > > > > > >name = avp_spec['/'(table_name|'$'db_scheme)] > > > >avp_spec = ''|'s:'|'i:'|avp_name|avp_alias > > > > > >Example 1-12. avp_db_load usage > > > >.... > >avp_db_load("$from","i:678"); > >avp_db_load("$ruri/domain","i:/domain_preferences"); > >avp_db_load("$uuid","s:404fwd/fwd_table"); > >avp_db_load("$ruri","i:123/$some_scheme"); > >.... > > > >---------------------------------------------------------------------------- > >----- > > > >My question are: > > > >Can someone explain the source syntax? > >Can someone explain the name syntax? > >what are i:, s:, avp_name and avp_alias? > > > >Thanks! > > > >Lenir > > > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)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 > > _____________________________________________________________________ > Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

18 years, 11 months

[Serusers] EmbeddedObj.o sipstack make error - Cannot compile b2bua

by administrator

Hello everybody, I am trying to install b2bus from cvs and I get the following error: make[1]: Leaving directory `/root/vocal/sip/sipstack' make[1]: Entering directory `/root/vocal/sip/sipstack' g++ -Wall -fPIC -Wno-deprecated -D_REENTRANT -DUSE_PTHREADS -g -I../../build -I../../build/../sdp/sdp2 -I../../build/../util -I../../build/../util/threads -I../../build/../util/logging -I../../build/../util/crypto -I../../build/../util/statistics -I../../build/../util/snmp -I../../build/../util/signals -I../../build/../util/behavior -I../../build/../util/io -I../../build/../util/services -I../../build/../util/transport -I../../build/../util/config -I../../build/../util/dnssrv -I../../build/../util/deprecated -I../../build/../util/adt -I../../build/../contrib/libxml2.Linux.i686 -I../../build/../contrib/libxml2.Linux.i686/include/libxml -I../../build/../contrib/libxml2.Linux.i686/include -DVOCAL_USE_DEPRECATED -DVOCAL_USING_PENTIUM -DOLD_PROV -c -o obj.debug.Linux.i686/EmbeddedObj.o EmbeddedObj.cxx BaseUrl.hxx:146: error: specialization of βtemplate<class _Key> struct __gnu_cxx::hashβ in different namespace /usr/lib/gcc/i386-redhat-linux/4.0.1/../../../../include/c++/4.0.1/ext/hash_ fun.h:71: error: from definition of βtemplate<class _Key> struct __gnu_cxx::hashβ EmbeddedObj.cxx: In member function βData Vocal::EmbeddedObj::doReverseEscape(const std::string&)β: EmbeddedObj.cxx:213: warning: comparison between signed and unsigned integer expressions make[1]: *** [obj.debug.Linux.i686/EmbeddedObj.o] Error 1 make[1]: Leaving directory `/root/vocal/sip/sipstack' make: *** [sip] Error 2 I user SER(0.9.4) and freeRadius (latest) on Fedora Core 4. Please Help ASAP. Thank you in advance Yiannis Marios

18 years, 11 months

[Serusers] EmbeddedObj Sipstack make error

by Yiannis Martis

Hello everybody, I am trying to install b2bus from cvs and I get the following error: make[1]: Leaving directory `/root/vocal/sip/sipstack' make[1]: Entering directory `/root/vocal/sip/sipstack' g++ -Wall -fPIC -Wno-deprecated -D_REENTRANT -DUSE_PTHREADS -g -I../../build -I../../build/../sdp/sdp2 -I../../build/../util -I../../build/../util/threads -I../../build/../util/logging -I../../build/../util/crypto -I../../build/../util/statistics -I../../build/../util/snmp -I../../build/../util/signals -I../../build/../util/behavior -I../../build/../util/io -I../../build/../util/services -I../../build/../util/transport -I../../build/../util/config -I../../build/../util/dnssrv -I../../build/../util/deprecated -I../../build/../util/adt -I../../build/../contrib/libxml2.Linux.i686 -I../../build/../contrib/libxml2.Linux.i686/include/libxml -I../../build/../contrib/libxml2.Linux.i686/include -DVOCAL_USE_DEPRECATED -DVOCAL_USING_PENTIUM -DOLD_PROV -c -o obj.debug.Linux.i686/EmbeddedObj.o EmbeddedObj.cxx BaseUrl.hxx:146: error: specialization of βtemplate<class _Key> struct __gnu_cxx::hashβ in different namespace /usr/lib/gcc/i386-redhat-linux/4.0.1/../../../../include/c++/4.0.1/ext/hash_fun.h:71: error: from definition of βtemplate<class _Key> struct __gnu_cxx::hashβ EmbeddedObj.cxx: In member function βData Vocal::EmbeddedObj::doReverseEscape(const std::string&)β: EmbeddedObj.cxx:213: warning: comparison between signed and unsigned integer expressions make[1]: *** [obj.debug.Linux.i686/EmbeddedObj.o] Error 1 make[1]: Leaving directory `/root/vocal/sip/sipstack' make: *** [sip] Error 2 I user SER(0.9.4) and freeRadius (latest) on Fedora Core 4. Please Help ASAP. Thank you in advance Yiannis Marios _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

18 years, 11 months

[Serusers] Error while compiling kphone package version 4.2

by rajesh.rajguru＠wipro.com

Hi, We are trying to make package of kphone from http://www.wirlab.net/kphone/ <http://www.wirlab.net/kphone/> We have downloaded kphone package version "KPhone 4.2" from the above mentioned site After referring to INSTALL instructions, we have executed following steps 1) go to kphone dir and do ./configure. This was executed properly 2) do make . When we are doing make we got following error ---------- make[1]: Entering directory `/home/wipro/SER_Test/kphone/kphone' c++ -I/usr/lib/qt3-gcc2.96/include -Wall -O3 -I. -I../gsm -I../ilbc -I../dissipa te2 -DHAVE_CONFIG_H -DSHARE_DIR=\"/usr/local/share/apps/kphone\" -DPO_DIR=\"/usr /local/share/kphone/translations//\" -c -o kphone.o kphone.cpp kphone.cpp: In member function `void KPhone::createTrayIcon()': kphone.cpp:272: no method `QPixmap::fromMimeSource' kphone.cpp:274: no method `QPixmap::fromMimeSource' kphone.cpp: In member function `void KPhone::updateTrayIcon()': kphone.cpp:297: no method `QPixmap::fromMimeSource' kphone.cpp:299: no method `QPixmap::fromMimeSource' make[1]: *** [kphone.o] Error 1 make[1]: Leaving directory `/home/wipro/SER_Test/kphone/kphone' make: *** [all] Error 2 ----------- In fact we also got this error for older versions of kphone package like KPhone 4.1.1, KPhone 4.1.0 We tried commenting call to fromMimeSource in file kphone.cpp, and then doing make, but it gives some unresolved symbols at the end Please let us know if you have also faced same types of errors and if yes then any pointers about this will be of great help Thanks in advance for your help Regards, Rajesh Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

18 years, 11 months

[Users] Beer on Wednesday 26th

by Helge Waastad

Hi, If I can make it, I'll stop by. The question is, what do SER/OpenSER people look like ;-) BTW, There will be a seminar, first of all regarding Asterisk but also discussing SER, on the 1th of November I guess it would be possible to have some discussions after this aswell. http://www.asterisk.no/asterisk/endelig_norges_foerste_asterisk_seminar_ meld_deg_paa_naa hw

18 years, 11 months

[Serusers] FW: Call logs

by Colin Jordan

I can not find where the call logs are in the ser database. What can I do to enable call information logging (time stamps, destination, exc) ? DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error please accept our apologies and notify the sender immediately. You must also delete the original message from your machine. If you are not the intended recipient, any use, disclosure, copying, distribution or action taken in reliance of it, is prohibited and maybe unlawful.

18 years, 11 months

[Serusers] fee-for-service SER server

by Michael Kendall

Dear Wizards, Many of you seem to be running public SER servers with hundreds of users. I would like to become a paying customer of such a SER server, but cannot find any such services advertised on the internet. Can someone please recommend a running, fee-for-service SER server? A few weeks ago, I asked this same question, and got zero responses. And I mean no responses of any kind. Is this the wrong forum to ask such a question? Even messages pointing out why I am a total idiot would have been preferable to total silence. If there are reasons that it is not practical for anyone to operate a fee-for-service SER server, I would love to see some discussion of what makes it so impractical. Please feel free to speculate on why you think such services have not been created. thanks, Michael

18 years, 11 months

[Serusers] latest rtpproxy + ser-0.8.14 + nathelper!!

by s0914508

Hi! I have some problems with rtp proxy. And the syslog tells that my rtpproxy can work (enable!!) ---------------------------------------- UA1: 10.10.54.131 (caller), behind NAT UA2: 218.171.144.86 (callee), public Internet SIP UA: Windows Messenger 5.1 SIP Server: ser-v0.8.14 Load module: nathelper RTP Proxy: latest rtpproxy (not portaone) NAT: CiscoSymmetric NAT ----------------------------------------- [Problem] Callee can hear caller's voice, but caller can't hear callee's. The the packet analysis is like this http://sip.csie.ncyu.edu.tw/~coolboy/calleereceiveinvite.JPG the line ( c= IN IP4 140.130.201.15 ) is NAT Server's IP address, not rtp proxy's IP address(140.130.175.2). I use force_rtp_proxy("140.130.175.2" ); to change c=IP4 with IP address(140.130.175.2), but it seems doesn't work. ----------------------------------------- Does anyone know how to solve my problem and the reason? It's very important to me. Thanks a lot !! ----------------------------------------- [Question] 1. Does c=IP4 get from rtpproxy if I use rtpproxy.?? And how they operate?? 2. I want to compile the latest nathelper module with FreeBSD system. I've read the 'Makefile', but it says "don't run with this Makefile, you should use 'master.Makefile' ". Does anyone can tell me how to do?? --== Mail Via Ncyu Mail ==--

18 years, 11 months

[Serusers] How to modify the URI.

by gc

I am new to SER. I'd like to modify some of the sip headers like 'uri' and 'from' header to remove +1 prefix from the phone number. What I want to do is to change +15042339988(a)3.2.4.3 to 5042339988(a)3.2.4.3 Can somebody show me how to do it in ser.cfg? gc

18 years, 11 months

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sr-users October 2005