sr-users July 2008

sr-users@lists.kamailio.org

147 participants
206 discussions

by Fero Vlkolinsky

Hello, I am getting the folowing error on communication with one of our application servers: tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 404 This is in situation where I have two triggerings to the AS (orig and term) and term subscriber is not registered to OpenSER. OpenSER thus returns 404 not found which is then sent back to OpenSER to be forwarded to orig leg of the AS. I had the very same behavior with other AS in our lab and found out that it was because of tag added to To header in AS for the 404 response. So i removed the tag and the call worked fine. However with this AS it doesn't seem to work. Is there a way to hack pick_branch so that I see what are the existing branches and why it does not match any? Or is there any easier way to find out the reason? Thanks a lot, Fero --- reklama ----------------------------------------------------- Hľadáš DOM – BYT – POZEMOK – KANCELÁRIU ? http://www.reality.sk

15 years, 8 months

[Kamailio-Users] Add a table to openser DB

by Diego Costantini

Hi, I need to use something like this (openser on a pc with 2 network cards): http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy since I couldn't find any documentation to properly create a "location_external" table, I followed this procedure: http://www.tutorialspoint.com/mysql/mysql-clone-tables.htm Now I have the table I need, but when I start openser I get this error: Jul 30 15:49:32 [8374] ERROR:usrloc:register_udomain: Invalid table version (use openser_mysql.sh reinstall) Jul 30 15:49:32 [8374] ERROR:registrar:domain_fixup: failed to register domain Jul 30 15:49:32 [8374] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 451 Jul 30 15:49:32 [8374] CRITICAL:core:fix_expr: fix_actions error Jul 30 15:49:32 [8374] ERROR:core:main: failed to fix configuration with err code -1 Jul 30 15:49:32 [8374] NOTICE:presence:destroy: destroy module ... How can I make Openser happy to accept my new table? Diego

15 years, 8 months

[Kamailio-Users] Getting SNMPStats to work

by Alex Massover

Hello! I'm trying to setup SNMPStats on Debian Sid. I have: dev5:~# cat /etc/openser/openser.cfg | grep snmp loadmodule "snmpstats.so" modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "MsgQueueMinorThreshold", 1234) modparam("snmpstats", "MsgQueueMajorThreshold", 5678) modparam("snmpstats", "dlg_minor_threshold", 500) modparam("snmpstats", "dlg_minor_threshold", 600) dev5:~# cat /etc/snmp/snmpd.conf | grep agent master agentx agentXSocket tcp:localhost:705 dev5:~# cat /etc/snmp/snmpstats.conf agentXSocket tcp:localhost:705 dev5:~# snmpd -f -D -Dagentx -x tcp:localhost:705 2>&1 | less No log handling enabled - turning on stderr logging registered debug token agentx, 1 netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() agentx_register_app_config_handler: registering .conf token for "agentxsocket" agentx_register_app_config_handler: registering .conf token for "agentxperms" agentx_register_app_config_handler: registering .conf token for "agentxRetries" agentx_register_app_config_handler: registering .conf token for "agentxTimeout" /etc/snmp/snmpd.conf: line 115: Warning: Unknown token: system.sysDescr.0. Turning on AgentX master support. agentx/config: port spec: tcp:localhost:705 agentx/master: initializing... agentx/master: initializing... DONE NET-SNMP version 5.4.1 agentx/master: unexpected callback op 4 agentx/master: handle pdu (req=0x365b5eac,trans=0x0,sess=0x0) agentx/master: open 0x86b37e8 agentx/master: opened 0x86c2028 = 6 with flags = a0 agentx/master: send response, stat 0 (req=0x365b5eac,trans=0x0,sess=0x0) But it doesn't work: dev5:~# snmpget -v2c -c public localhost openserSIPEntityType.0 OPENSER-SIP-COMMON-MIB::openserSIPEntityType.0 = No Such Object available on this agent at this OID Am I missing something? -- Best Regards, Alex Massover

15 years, 8 months

[Kamailio-Users] OpenSER + Asterisk (Voicemail) Issue....

by Gerard A. Matthew

I'm having this little issue when implementing the voicemail feature. My openser.cfg looks like this in the failure route: if(!t_was_cancelled()) { revert_uri(); rewritehostport("voicemail.mydomain.com:5061"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } On my asterisk end after the time out, i'm viewing the following: SELECT * FROM sipusers WHERE name = 'XXXXXX' i.e XXXXXX = the PSTN number i'm using to call into the IP phone that's connected to OpenSER There seems to be a simple mixup with the number that is sent to asterisk. Obviously there is no user with the PSTN number, however there is one with the called number. Any idea as to what wold be causing this? Have I provided enough information? Regards, Gerard.

15 years, 8 months

[OpenSER-Users] Polycom Phones produce no wringing when dialing pots lines on the pstn

by Alex G

I'm sure i'm not the first to experience this. My polycom phones when calling out a pstn gateway recieve far end ringing when dialing cellphones in the US, but when i try to call a pots line i get no ringing, only silence until the end point answers the call. Does anyone have a solution for this? From what i can see in the sip captures, the gateway sends a 180 ringing on the pots line calls, but when i call a cellphone i recieve a 183 Session Progress where the phones do provide the ringing. A polycom to polycom call shows the same 180 ringing, but in this case they do provide ringing. Just to be clear, the ringing i am talking about is the rinigng you here in the earpiece after a dial. Any insight or help would be very appreciated. I have sip captures that i can post, but opted not to incase someone has seen this and knows the answer first. Thanks so much!

15 years, 8 months

Re: [Serusers] use_dns_cache and call forwarding

by caio

Hi Miklos, This happens only when I try to do a call to a subcriber having forwarding enabled. Normal calls success ok. I'm not sure about the append_branch() usage and how to work with 'on busy' and 'no answer' forwarding features. Thanks your comments.. Claudio On Thu, Jul 24, 2008 at 4:58 AM, Miklos Tirpak <miklos(a)iptel.org> wrote: > Hi, > > On 07/23/2008 05:22 PM, caio wrote: >> >> Hi guys, >> I have a problem with call forwarding (on busy or not answer).., the >> call fall into the failure_route(1) block, and AVP checking and pushto >> does its job. >> >> Here a snipped of code of failure_route(1) (flag 27 means call fwd if >> no answer): >> >> if (isflagset(27) && t_check_status("408")) { # if fwd no >> answer is set and reply msg is 408 >> >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> resetflag(27); >> >> avp_print(); >> >> log(1, "LOG: --> appending new branch..."); >> if (!append_branch()){ >> t_reply("500", "Too many branches?!"); >> drop; >> } >> >> log(1, "LOG: --> Calling route_6..."); >> route(6); >> break; >> >> Route(6) finally calls route(1), where t_relay is called. >> But have a dns failure..and the call never is forwarded. >> >> == ser.log == >> >> DEBUG: mk_proxy: doing DNS lookup... >> get_record: lookup(_sip._udp.wsa.lab, 33) failed_ >> sip_resolvehost: no SRV record found for wsa.lab, trying 'normal' >> lookup..._ >> check_via_address(190.244.33.5, 190.244.33.5, 0)_ >> ERROR: udp_send: sendto(sock,0x2ab32599c720,910,0,0x2ab325999e68,16): >> Operation not permitted(1)_ >> msg_send: ERROR: udp_send failed_ >> ERROR: t_forward_nonack: sending request failed_ >> DEBUG: add_to_tail_of_timer[4]: 0x2ab325999e80_ >> DEBUG: add_to_tail_of_timer[0]: 0x2ab325999ea0_ >> ERROR: w_t_relay (failure mode): forwarding failed_ >> >> A question regarding this trouble.....Can I disable use_dns_cache ? >> SER version is 0.9.7. > > there is no DNS cache in version 0.9.x as I recall, but anyway, your problem > does not seem to be related to DNS, but to the UDP message sending: SER > fails to send out the UDP packet, "Operation not permitted". Is there no > firewall or routing problem at the proxy side? > > Miklos > >> What do you recommend? >> If you need my ser.cfg I can paste it... >> >> Thanks.. >> > -- caio -- caio

15 years, 8 months

[Kamailio-Users] rtpproxy scalability.

by Alex Balashov

Does anyone have any reliable data on rtpproxy dimensioning, i.e. just about how many calls per box (and what kind of specs) it can really handle in actually-existing practice, not in theory? The one time I used it for NAT traversal was on a pretty beefy quad-CPU box, and I started seeing clicks and distortions in the calls beyond a load of 90 or so. However, I was never able to decisively ascertain that the cause is not the network. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

15 years, 8 months

[Kamailio-Users] new list for announcements

by Henning Westerholt

The Kamailio project has now a dedicated list for announcements. There will be only important messages posted to this list, like release announcements, security advisories, informations about important changes in the project organisation and community events. The posting frequency will be problably not more then a few per month. All postings will be moderated. The address of the list is: kamailio-announce(a)lists.kamailio.org, to subscribe visit: http://lists.kamailio.org/cgi-bin/mailman/listinfo/kamailio-announce Best regards, Henning

15 years, 8 months

Re: [Kamailio-Users] Prepaid system using openSER

by Darren Sessions

I’ve used SER and now OpenSER (uhhm, Kamailio) for the sip proxy and cdr generation (along with media proxy and cdrtool) and Asterisk as the ivr piece for years. Using a proxy like OpenSER/Kamailio to front your IVR servers is your best bet. The combination has worked extremely well for me. - Darren -- www.darrensessions.com On 7/28/08 6:52 PM, "Rana Dhekial" <dhekial(a)msn.com> wrote: > Hi, > > We have a requirement to build a prepaid system using OpenSER. I > took a look at CDRTool which looks quite promising. But I am not > sure how to build the Call control module and provide IVR type > information to the end users. > > If any one has done similar type of job, please respond. > > thanks, > > Rana > > Keep your kids safer online with Windows Live Family Safety. Help > protect your kids. <http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa… > >

15 years, 8 months

[Kamailio-Users] HELP: Create new table for Nathelper

by Diego Costantini

Hi, I repropose my unanswered question. I was following this sample configuration for a multihomed server: http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy and it mentions: "For this example to work, you must also follow the nathelper module instructions for creating new location_internal and location_external tables. Be careful to use underscores ("_") and not hyphens ("-") when creating the new tables." But I couldn't find any instruction anywhere. Can someone address me to the correct place please? Thanks, Diego

15 years, 8 months

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sr-users July 2008