sr-users April 2013

sr-users@lists.kamailio.org

94 participants
132 discussions

Abort at qm_debug_frag function
by Bruno Bresciani 03 Apr '13

03 Apr '13

Kamailio is generating a abort at qm_debug_frag function (BUG: qm_*: prev. fragm. tail overwritten) but I can't understanding the meaning this abort... Someone can tell me the purpose of qm_debug_frag function, it checks if some improper memory handling happened and abort the program? Below is backtrace of abort: #0 0x00f41402 in __kernel_vsyscall () #1 0x009fec10 in raise () from /lib/libc.so.6 #2 0x00a00521 in abort () from /lib/libc.so.6 #3 0x081797b7 in qm_debug_frag (qm=0xb5f83000, f=<value optimized out>) at mem/q_malloc.c:158 #4 0x0817a6f3 in qm_free (qm=0xb5f83000, p=0xb6109a54, file=0x43760e "perms_db: src_ip.c", func=0x4378a0 "list_destroy", line=79) at mem/q_malloc.c:442 #5 0x0043563c in list_destroy (liste=0xb6107274) at src_ip.c:79 #6 0x00430a93 in reload_srcip_table () at db.c:163 #7 0x004313be in perms_db_srcip_reload (cmd_tree=0x835aeac, param=0x0) at fifo.c:107 #8 0x00335d97 in mi_fifo_server (fifo_stream=0x99561d8) at ../../lib/kmi/mi.h:77 #9 0x00337621 in fifo_process (rank=1) at mi_fifo.c:235 #10 0x003378a5 in mi_child_init (rank=0) at mi_fifo.c:199 #11 0x08120a02 in init_mod_child (m=0x82f4b88, rank=0) at sr_module.c:829 #12 0x081209dc in init_mod_child (m=0x82f4d1c, rank=0) at sr_module.c:826 #13 0x081209dc in init_mod_child (m=0x82f5164, rank=0) at sr_module.c:826 #14 0x081209dc in init_mod_child (m=0x82f79f4, rank=0) at sr_module.c:826 #15 0x081209dc in init_mod_child (m=0x82f7cf0, rank=0) at sr_module.c:826 #16 0x081209dc in init_mod_child (m=0x82f8174, rank=0) at sr_module.c:826 #17 0x081209dc in init_mod_child (m=0x82f8798, rank=0) at sr_module.c:826 #18 0x081209dc in init_mod_child (m=0x82f8b54, rank=0) at sr_module.c:826 #19 0x081209dc in init_mod_child (m=0x82f8eb4, rank=0) at sr_module.c:826 #20 0x081209dc in init_mod_child (m=0x82f90b4, rank=0) at sr_module.c:826 #21 0x081209dc in init_mod_child (m=0x82f952c, rank=0) at sr_module.c:826 #22 0x081209dc in init_mod_child (m=0x82fa980, rank=0) at sr_module.c:826 #23 0x081209dc in init_mod_child (m=0x82fab9c, rank=0) at sr_module.c:826 #24 0x081209dc in init_mod_child (m=0x82faff8, rank=0) at sr_module.c:826 #25 0x080ae175 in main_loop () at main.c:1624 #26 0x080b1256 in main (argc=3, argv=0xbf9a8f14) at main.c:2398 Best Regards

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Re: [SR-Users] sr-users Digest, Vol 95, Issue 17
by julian arsanches 03 Apr '13

03 Apr '13

Thanks a lot it was just what i need it , it worked like a charm. strip(3); On Wed, Apr 3, 2013 at 3:39 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: please help with some transformations. (Alex Balashov) > 2. Re: please help with some transformations. > (Daniel-Constantin Mierla) > 3. Abort at qm_debug_frag function (Bruno Bresciani) > 4. Re: Abort at qm_debug_frag function (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 03 Apr 2013 14:18:18 -0400 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] please help with some transformations. > To: sr-users(a)lists.sip-router.org > Message-ID: <515C726A.10104(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello Julian, > > On 04/03/2013 01:54 PM, julian arsanches wrote: > > > Hi am new to kamilio , but able to make calls, i need to do something > > really simple if someone can give me an example of it i will be > gratefull. > > > > just before the call gets relayed i need to delete a prefix from the sip > > uri so it gets transformed to the same uir but less 3 digits. > > > > for example sip:123xxx@domain.com <mailto:sip%3A123xxx@domain.com> >>> > > changed to sip:xxxx@dmain.com <mailto:sip%3Axxxx@dmain.com> > > > > i tryed with transformations but still too new and does give me so many > > errors that kamailio don't start, > > There is a core convenience function that can help with this, as long as > you're manipulating the request URI: > > strip(3); > > You referred to "the SIP URI", which, of course, is ambiguous since a > SIP request has many SIP URIs. > > -- Alex > > -- > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > United States > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > > > > ------------------------------ > > Message: 2 > Date: Wed, 03 Apr 2013 20:18:44 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] please help with some transformations. > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <515C7284.5060905(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > if you just need to remove first 3 digits, use: > > strip(3); > > If the number of digits to strip is variables, use strip transformation: > > - http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#sstrip_len > > If does not work for you giving errors, paste here what you have in the > config for that part. > > Cheers, > Daniel > > On 4/3/13 7:54 PM, julian arsanches wrote: > > Hi am new to kamilio , but able to make calls, i need to do something > > really simple if someone can give me an example of it i will be > > gratefull. > > > > just before the call gets relayed i need to delete a prefix from the > > sip uri so it gets transformed to the same uir but less 3 digits. > > > > for example sip:123xxx@domain.com > > <mailto:sip%3A123xxx@domain.com> >>> changed to sip:xxxx@dmain.com > > <mailto:sip%3Axxxx@dmain.com> > > > > i tryed with transformations but still too new and does give me so > > many errors that kamailio don't start, > > > > tryed subs_uri but i dont know how to tell it to just take the first > > tree digits, it is usually done for adding prefixes. > > > > please help. > > > > thank you in advance. > > > > on my scenario i am having an asterisk that i dont control sending me > > an invite for with a prefix i need kamailio to forward that uri to > > another server minus the prefix. > > > > thank you. > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > >

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please help with some transformations.
by julian arsanches 03 Apr '13

03 Apr '13

Hi am new to kamilio , but able to make calls, i need to do something really simple if someone can give me an example of it i will be gratefull. just before the call gets relayed i need to delete a prefix from the sip uri so it gets transformed to the same uir but less 3 digits. for example sip:123xxx@domain.com >>> changed to sip:xxxx@dmain.com i tryed with transformations but still too new and does give me so many errors that kamailio don't start, tryed subs_uri but i dont know how to tell it to just take the first tree digits, it is usually done for adding prefixes. please help. thank you in advance. on my scenario i am having an asterisk that i dont control sending me an invite for with a prefix i need kamailio to forward that uri to another server minus the prefix. thank you.

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Kamailio World Updates - April 3
by Daniel-Constantin Mierla 03 Apr '13

03 Apr '13

Hello, just a bit less than two weeks to the start of Kamailio World Conference, we are gearing up and already started preparing everything for the event. The agenda with presentations is available at: - http://conference.kamailio.com/k01/schedule/ Details about the speakers are available at: - http://conference.kamailio.com/k01/speakers/ Regarding Kamailio project, even there was an extended weekend with public holidays in around Easter, three new modules showed up - stun, dnssec and sipt - and a lot of new code made its way to old modules, among most active being snmpstats. If you haven't registered yet for the Kamailio World conference, be aware of that standard registration ends this week, on April 5, 2013. Actually the registration might be completely closed as it is very likely that the event will get full booked, even we extended the capacity about one week ago. It is a great opportunity to meet other hundreds people from real time communications services. Do not delay registration if you plan to attend the event, the form and more details at: - http://conference.kamailio.com/k01/registration/ Several companies coming with stands for presenting products, services and demos will welcome you to stop by just in front of the conference room - Sipwise, Sipgate, NG Voice, FhG Fokus and Asipto. Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Evariste Systems to drop Kamailio market, become Acme Packet VAR
by Alex Balashov 02 Apr '13

02 Apr '13

For immediate release: ATLANTA, GA (1 April 2013)--Evariste Systems LLC, an Atlanta-based consultancy specialising in Kamailio-based VoIP infrastructure solutions for the ITSP and CLEC market, has announced that beginning in the second quarter of 2013, it will be abandoning its Kamailio-based technology portfolio to focus on its new role as a preferred VAR (Value Added Reseller) for Acme Packet (NASDAQ:APKT). "It is with a heavy heart that we abandon five years of Kamailio-oriented work and the Canonical SIP Routing Platform product derived from it," said Alex Balashov, the principal of the company. "However, the reality is that investment in open-source VoIP technology is a dead end. From a technological point of view, we have lagged very badly in meeting the needs of today's sophisticated VoIP market, and it's time to cut our losses. Asterisk, Kamailio, FreeSWITCH--all this stuff just hasn't kept up with the pace of evolution of 3GPP, ETSI, and ITU standards. We are tired of saying 'sorry, we don't support IMS or H.323' to our resultingly dwindling customer base. Does anyone actually run an all-SIP network?" Starting in early April, Evariste will begin providing value-added consultancy related to the implementation of the Acme Packet Net-Net Session Director. In Balashov's view, "the Net-Net SD is the only product capable of meeting the perimeter security, routing and peering needs of today's VoIP service delivery environment." Fred Posner, the director of Team Forrest, a Palner Group integration and consultancy operation based in the Jacksonville, Florida area, agreed: "SIP is a tiny piece of the telephony puzzle. The big boys of ClueCon [an interoperator revenue-sharing consortium] want DIAMETER-based interdomain peering policy control, H.323, MGCP, and IMS. IMS is pretty much how VoIP architecture is done now. We got out of the Asterisk business just in time, right before Mitel swallowed the PBX world. I'm glad to see Evariste is finally seeing the light, and I'm sure its shareholders are too." Posner also believes Evariste's lack of support for TDM interfaces accounted for dwindling market share. "Have you seen CSRP? It's SIP in, SIP out. Real inter-LATA haulers and application service providers use TDM and leave SIP for things like voicemail. I can't plug my DS3s into a SIP proxy, so I just don't think there was any real demand for the sort of thing they were doing." Noting Oracle's US$2.5bn acquisition of Acme Packet in early February, as well as its more recently announced buyout of Tekelec, a Siris Capital Group portfolio company, Balashov remarked: "The obvious shift to an Oracle-centric telephony paradigm was a kind of validation, if you will, of our decision to unload our dead weight and sign on to the revolution in unified communications." Sean McCord, of CyCORE Systems, an Atlanta-based software consulting house and long-time Evariste creditor, agreed that there was a natural synergy between Evariste's shift to Acme Packet and Oracle's dominance of telephony infrastructure. "Oracle is a forward-thinking telecom pioneer," McCord said. "The telephone is Oracle, and Oracle is the telephone." Balashov also noted that a tightening regulatory environment and new consumer protection rules helped hasten the decision to embrace the more professionalised Acme Packet product portfolio. John Knight, Senior Engineer at Hendersonville, NC-based Ringfree Communications, one of Evariste's oldest channel partners, said: "As one of Evariste's long-time disties, we were jittery about exposure to CALEA and the QA requirements of large call centers. We tried to make do, but at some point we just had to put the relationship on stop. I'm all in favour of open, but there's just no open-source software out there that does call recording, and that's the bottom line for us. In the end, we had to restructure some debt just to get bondholders to let us source a proprietary solution on tick." In a thematically related move, Evariste will be dropping its heavy use of the open-source PostgreSQL database manager for its rating and reporting tools. "The business case for standardising on Oracle's databases could not be clearer. With Oracle Database 11g's support of warehousing and OLTP, the real mystery is why we didn't go there sooner," said Balashov. Carlos Alvarez, a director at Televolve, a growing Phoenix-area VoIP operator, recently spearheaded a move away from Evariste's PostgreSQL- based call detail record (CDR) storage solution to one running atop Microsoft SQL Server 2008. Alvarez commented: "Evariste had a nice idea, in a cute, David-and-Goliath kind of way, but we're processing over five hundred phone calls a day now. Are we really going to store those kinds of volumes in an open-source database? Might as well just put it all in flat text files at that point. Phone service is an uptime game. You can't compromise on this stuff. What if someone needs to call 911?" Asked to summarise his expectations, Balashov said: "I hope this turns us around in a big way. We were wrong to think that nobody cared about stuff like P-CSCFs, or that you could deliver even rudimentary VoIP to the premise without the expansive feature set of a comprehensive solution like the Net-Net SBC. I can only hope the market forgives us for betting on 'SIP Express Router' and its ilk back in the day, and gives us a chance to do it right in round two." Fred Posner, of Team Forrest, added: "Besides, if you look at the Git repository, Kamailio hasn't had any code contributions in at least five years. It seems everyone's figured out this pure SIP stuff is defunct and hokey."

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ClueCon 2013 - Call For Speakers
by Michael Collins 02 Apr '13

02 Apr '13

[image: https://mail-attachment.googleusercontent.com/attachment/u/0/?ui=2&ik=43fec…]<http://www.cluecon.com> ClueCon - the open source IP communications conference by developers, fordevelopers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea fora technical presentation for ClueCon 2013 then we would like to hear about it. What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon *loves *technical presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: - ClueCon talks are 30 minutes in length, including Q&A time with the audience - ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio - Attendees enjoy hearing about projects built with open source tools, especially those in a production environment - Highly technical discussions that show the nuts and bolts are especially well-liked - The audience appreciates seeing and participating in live demonstrations - We are especially interested in WebRTC-related talks and demonstrations Please send your proposals to marketing(a)cluecon.com. Be sure to include the following items: - Working title - Brief description of the talk (abstract) - Name of the presenter Don't delay! There are a limited number of openings. We will contact you as soon as your talk has been approved and will inform you of the scheduled time. ClueCon 2013 Registration Information ClueCon 2013 registration is now open!. Visit the registration page<http://www.cluecon.com/register?cc12cfs> for details. Be sure to book your room at the Hyatt Chicago Magnificent Mile<http://www.cluecon.com/hotels/>and qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2013. Also, keep in mind that the FreeSWITCH community has a conference call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2013. Stay tuned for more news about ClueCon speakers, sponsors, and related events! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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alphabetic indexes generated for development version
by Daniel-Constantin Mierla 02 Apr '13

02 Apr '13

Hello, I generated the alphabetic indexes for development version, there are already new modules and code since release of 4.0. If anyone spots missing elements (mod params, functions, etc...), write back here so we fix the issue (could be wrong formatted xml docbook file for module readme). They are available at: - http://www.kamailio.org/wiki/#alphabetic_indexes Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Are there any Blind Call Forwarding examples available for Kamailia 3.3.0?
by alans126 02 Apr '13

02 Apr '13

I've just configured Kamailia and users 10001,10002,10003 can give each other a call successfully. Now I want to forward every call to 10002 to 10003. The only example I can find is for OpenSER 1.X, the configure file seems different from Kamailia 3.3's. I tried to add the following code to the configure file and some other add_module parts to the other one, but it doesn't work... if(avp_db_load("$ru","$avp(s:callfwd)")) { $ru = $avp(s:callfwd); route(RELAY); exit; }. Can anyone provide me with the kamailia.cfg and kamctlrc files which implemented call forwarding unde Kamailia 3.3, please? I'm a newbie so I really hope to get your help. 2013-04-02 alans126

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Support
by Maritza y Cristian Zuÿfffff1iga 02 Apr '13

02 Apr '13

Dear Sr, Is it possible to use QuteCom softphone with iptel.org SIP service? Regards Cristian A. Zúñiga

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Is cvs repository of sip_route still alive?
by Yuming Zheng 02 Apr '13

02 Apr '13

Hi, export CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/ser cvs login OK, when try checkout the sip_route cvs co sip_router cvs [checkout aborted]: cannot stat /cvsroot/ser/locks: No such file or directory Although cvs view is working. I want go check some logs in the repository. BR, Frank.zheng

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sr-users April 2013