sr-users February 2007

sr-users@lists.kamailio.org

158 participants
285 discussions

by Hansi Lao

Hello All, I just recently migrated to openser 1.2 from 1.1.x. how can i migrate my database from mysql? do i have to recreate the whole database again? or is there another alternative? Openser halts since its encounters this error which is obviously related to the database backend: 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) ERROR:auth_db:auth_fixup: Invalid table version (use openser_mysql.sh reinstall) 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 122 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration 0(0) INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill Any help wil be greatly appreciated. Thanks!

17 years, 9 months

[Users] 1.2.0 and retransmissions

by Helge Waastad

17 years, 9 months

[Serusers] Prefix from the To user to the from user

by Ibrahim Hamouda

Hi all This is my first post to the list, so I hope I can get some help. Here is the scenario. I'm building a SER server 0.9.6 with the following characteristics. I'll be supporting different companies with multiple locations each. The phone number will be 7 digits. First 2 digits are the company, the following 2 digits are the branch or location and the last 3 digits are the user number. So, let's say companyA digits are 21, and companyB digits are 22. CompanyA has two branches 10 and 11. Branch 10 has a user 100 (full number in subscriber table is 2110100), and a user 101 (full number in subscriber table is 2110101) Branch 11 has a user 100 (full number in subscriber table is 2111100). CompanyB has a branch 10 and user within the branch 100 (full number in subscriber table is 2210100) I want user when dialing within the same branch to be able to dial the other user number only. User 2110100 can dial 100 to reach User 2110101. But if he wants to dial User 2111100 (Same company different branch) he can dial 11100. And if he wants to dial a user within another company he has to dial the full 7 digits. No to SER config If ser receives a 3 digits number in the $ruri, it should extract the first 4 digits from the From URI and prefix the $ruri/username with them. If ser receives a 5 digits number in the $ruri, it should extract the first 2 digits from the From URI and prefix the $ruri/username with them. My problem is how to extract the first 2 or first 4 digits from the from URI. Any ideas? Thanks in advanced for your help, I have spent 5 days now trying to figure this one out. Ibrahim Hamouda

17 years, 9 months

[Users] NAT Problem

by raviprakash sunkara

Hello Users, Good Morning, What are Type of the NAT is used for OpenSER for production , OpenSER is inside the NAT (router/firewall ) 1) I tested with 2 SJ phones and 2 cisco ATA phones, UAC's are in Behind the NAT, UACs are in the Same N/w with one public_address, SJ phone show the NAT type is Symmetric NAT this case is Working Fine, For example : [Behind the NAT1 ] [Behind the NAT2] [Behind the NAT1 ] SJphone --------| firewall]-----------------------> OpenSER-------[firewall]--------------> ATA phone 2) When SJ phones or HardPhones are in Different network behind the NAT of the OpenSER server, here SJ phone shows the Port Restricted core NAT Here media is not is signaling at all , in this Case.... For example : [Behind the NAT1 ] [Behind the NAT2] [Behind the NAT3 ] SJphone ----------[firewall]---------------------> OpenSER--------[firewall ]-------------> ATA phone SIP NAT Traversal support only Symmetric NAT , ? Then How to solve this issue ? Please Help ? -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

17 years, 9 months

[Serusers] Linksys WRT54GL w/DD-WRT voip (SIPatH) & external registration

by Bruce

Does anybody has the experience with the Linksys router WRT54GL running DD-WRT v23 SP2 (09/15/06) voip (SIPatH) firmware? I would like to register one external phone at another registrar through the router (using Outbound proxy). I have modified the ser.cfg to process external requests: if (method=="REGISTER") { if (src_ip==192.168.0.0/16 or src_ip==172.16.0.0/12 or src_ip==10.0.0.0/8 or src_ip==X.Y.0.0/16) where X.Y.z.t is the WAN IP address of the SIP Phone on the Internet but unfortunately I have discovered the router is dropping incoming external SIP packets and not passing them to the SIPatH server for processing. Processing of local LAN SIP requests works fine. The router has the LAN address 10.0.0.138. Forwarding the port 5060 to 10.0.0.138 does not help. Any advice would be highly appreciated. Main goal is to allow external phone, registered at another registrar trough the router, to call local LAN SIP phones (and back) trough the SIPatH.

17 years, 9 months

[Users] IRC channel: #openser

by Daniel-Constantin Mierla

Hello, this should be interesting for IRC fans. Many openser folks are hanging around #openser channel at irc.freenode.net. It should be a good place to shoot quick questions, answers may get back instantly. Some details: http://www.openser.org/index.php?option=com_content&task=view&id=86&Itemid=… I strongly recommend that important issues/solutions should be summarized to mailing lists. The lists are archived online, improving the knowledge base publicly available -- new comers will have access to it and therefore the learning process is speed up. The channel is in the process of getting officially reserved and assigned to OpenSER project. Many thanks to Adam Linford and all others spending time to get this done. Have fun there, Daniel

17 years, 9 months

[Users] Sip User Agent and IPv6 with OpenSER 1.1.1

by pat.kush＠verizon.com

Hello, I am setting up an end-to-end IPv6 testbed at Verizon which includes PCs with IPv6-related applications, Servers, Edge Routers, and Backbone Routers. Currenly I am looking for a SIP client (SIP User Agent) for IPv6 that works on VISTA, XP, or Linux , and interoperates with OpenSER 1.1.1. Does anyone know of or has a SIP client that works on IPv6 networks with OpenSER 1.1.1? Has anyone used OpenSER 1.1.1 for IPv6? Is there any documentation for this, or bugs ? Thanks. Pat Kush Pat Kush Distinguished Member Technical Staff Verizon Technologies 40 Sylvan Road Waltham, MA 02451

17 years, 9 months

[Serusers] uac module in 9.6

by Stephen Helfen

I am under the impression that 9.6 can be run including the uac module in order to implement the uac_replace_from function. Have I been misinformed?Thanks,Steve _________________________________________________________________ Get the most out of tax shelters and other breaks. http://articles.moneycentral.msn.com/Taxes/TaxShelters/TaxShelters.aspx?ici…

17 years, 9 months

[Users] Re: [Devel] Sip User Agent and IPv6 with OpenSER 1.1.1

by pat.kush＠verizon.com

Thank you. We were in contact with Counterpath about 2 months ago. At that time Eyebeam did not support IPv6. We have used Eyebeam in the IPv4 portion of our IPv6 NGNLAB. Pat "Klaus Darilion" <klaus.mailinglists To: Patricia J. Kush/EMPL/MA/Verizon@VZNotes @pernau.at> cc: users(a)openser.org, team(a)openser.org, devel(a)openser.org Subject: Re: [Devel] Sip User Agent and IPv6 with OpenSER 1.1.1 02/16/2007 06:57 AM pat.kush(a)verizon.com wrote: > Hello, > > I am setting up an end-to-end IPv6 testbed at Verizon which includes PCs > with IPv6-related applications, Servers, Edge Routers, and Backbone > Routers. > > Currenly I am looking for a SIP client (SIP User Agent) for IPv6 that works > on VISTA, XP, or Linux , and interoperates with OpenSER 1.1.1. Counterpath says that eyebeam supports IPv6. regards klaus > > Does anyone know of or has a SIP client that works on IPv6 networks with > OpenSER 1.1.1? > > Has anyone used OpenSER 1.1.1 for IPv6? Is there any documentation for > this, or bugs ? > > Thanks. > > Pat Kush > > Pat Kush > Distinguished Member Technical Staff > Verizon Technologies > 40 Sylvan Road > Waltham, MA 02451 > > > > > _______________________________________________ > Devel mailing list > Devel(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/devel -- Klaus Darilion nic.at

17 years, 9 months

[Users] Openser and NATed users

by Victor Gamov

Hi guys! I try to use OpenSER with NATed-users and now I have some problem. No I'll try to explain it. I have the following network scheme: mySIP-phone_behind_NAT --> NAT --> OpenSER --> OtherProxy --> other_SIP_phone mySIP-phone_behind_NAT have some knowledges about external NAT_ip so it successfully registered on OpenSER using nathelper module and location table have new record looks like: user@mydomain Contact: user@NAT_ip:5060 Received: NAT_ip:NAT_port Then mySIP-phone_behind_NAT send new INVITE with Contact=user@NAT_ip:5060 OpenSER fixup this Contact using fix_nated_contact() and send INVITE to OtherProxy with Contact=user@NAT_ip:NAT_port;nat=yes Then 200OK and ACK successfully traversed via OpenSER and dialog esteblished. Then other_SIP_phone sends BYE request using R-URI learned from previous INVITE Contact-header=ser@NAT_ip:NAT_port;nat=yes OpenSER rewrite this R-URI to user@NAT_ip:NAT_port and send it using t_relay(). The packet goes to NAT_ip:NAT_port but mySIP-phone_behind_NAT sends "404 Not found" because it does not know anything about user@NAT_ip:NAT_port -- it expected user@NAT_ip:5060 as sended in first INVITE. So, my question is: is this my misconfiguration or incorrect logic understanding ? I have some ideas how to fix this problem, for eaxample: when I got INVITE from mySIP-phone_behind_NAT I must rewrite Contact to user@mydomain (assuming that requests to mydomain goes to OpenSER always). Then if OpenSER got BYE from other_SIP_phone it will lookup() and restore Contact and Received info. But I have problem again: If I rewrite R-URI to Contact and t_relay() it -- packet will be discarded by NAT because it does not anything about port 5060. If I rewrite R-URI using Received info I will get problem discribed early. Feature request "Allow AVP in t_relay()" (published in Tracker) will solve this problem Any suggestion? -- CU, Victor Gamov

17 years, 9 months

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sr-users February 2007