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From: OpenSBC F. <ope...@op...> - 2008-11-18 09:55:07
|
Hi Joegen, Seems I achieved some progress. Inbound calls place to a sip-trunk listener now. But I'm getting Internal Server Errors. Where is my mistake? 340834:04:20.473 INF: [CID=0x07a9] >>> SIP/2.0 500 Internal Server Error DST: 10.100.1.37:9000:UDP SRC: 10.100.1.37:9006 enc=0 bytes=443 340834:04:20.473 DBG: [CID=0x07a9] 340834:04:20.473 DBG: [CID=0x07a9] SIP/2.0 500 Internal Server Error 340834:04:20.473 DBG: [CID=0x07a9] From: <sip:413...@vo...>;tag=quz8d19vg2 340834:04:20.473 DBG: [CID=0x07a9] To: <sip:445...@vo...;sip-trunk=true>;tag=7c0ad6efc5b3dd118a28d5799cb4286a 340834:04:20.473 DBG: [CID=0x07a9] Via: SIP/2.0/UDP 10.100.1.37:9000;iid=3987;branch=z9hG4bK04c8c7efc5b3dd118a28d5799cb4286a;uas-addr=10.100.1.37;rport=9000;received=10.100.1.37 340834:04:20.473 DBG: [CID=0x07a9] CSeq: 1 INVITE 340834:04:20.473 DBG: [CID=0x07a9] Call-ID: 3c3a0337c788-s6smov9nh4bh-0x0015 340834:04:20.473 DBG: [CID=0x07a9] Server: OpenSIPStack-1.1.7-22 340834:04:20.473 DBG: [CID=0x07a9] Content-Length: 0 Thanks for help. |
From: OpenSBC F. <ope...@op...> - 2008-11-17 17:23:30
|
Hi Joegen, You are right. My OpenSBC is behind a NAT. Incoming requests look like: INVITE sip:41445555828@*my_gateway_ip*:9006;transport=udp SIP/2.0 State: 0 How can I make OpenSBC to send it over sip-trunk as incoming? I tried to add a B2BUA route {color:#000080}[sip:*@*my_gateway_ip*:*] sip:trunkname.org;sip-trunk=true{color} but it was not help. May be it is need to add another sip-trunk for incoming calls? Please suggest. |
From: OpenSBC F. <ope...@op...> - 2008-11-15 04:51:46
|
hi, i just want to ask regarding your setup, the way i understand it, is this: nic1 - public ip (with sipx sipX+DHCP+DNS) nic2 - private ip (with osbc) is this correct? thanks raymund |
From: OpenSBC F. <ope...@op...> - 2008-11-14 11:16:29
|
Hi Joegen, Seems I used double trunk which is hard to analise. It was call from 50...@my... => trunk to 41445001800 - provider sip.backbone.ch -> 41445001828 => trunk 52...@my... I just made another call: Analog phone 41448742528 => 41445001828 (provider number) => trunk to 528 and send log to your e-mail. Thank you |
From: Joegen E. B. <joe...@gm...> - 2008-11-14 03:52:24
|
Hi, I have seen the log. Form what I could tell, you are trying to place an inbound call to a sip-trunk via the main trunk sip listener. That will not be processed as an inbound trunk call. Instead it will be processed as outbound. So there is a screw up somewhere. Make sure to send inbound calls to a siptrunk via the sip-trunk listener. I am also not 100% clear what your setup is. If you could send a detailed call diagram of what you want to achieve, so much the better. Joegen OpenSBC Forum wrote: > I added B2BUA route [sip:100] sip:sipx.ebc.fleurop > > > Unfortunately it didn't help. I will send the logby mail. > > > Thanks for reply. > > > > |
From: OpenSBC F. <ope...@op...> - 2008-11-13 00:19:57
|
Greetings, I am trying to use OpenSBC with SipXecs for sip trunk registration and NAT traversal. I have them both on a single box and am trying to get the OpenSBC part to work with sipX. Reading the OSBC and sipX forums has left me a bit confused about a few things, such as: 1) I haven't been able to get the sip trunks to register, and judging from the log file, they don't even attempt to register. If I have a xml configuration in the SIP-Trunk-config xml window, can I expect it to attempt to register even if it's wrong? or do I need to do something to tell it to attempt to register? Does the fact that mine does not even attempt to register mean that something else that is wrong is keeping it from attempting to register? 2) Should I be using Upper Registration? or some other mode with SipX? 3) Should the B2BUA routes go both ways, one set toward the sipx box and another toward the sip provider? 4) Should UpperReg routes (if needed) go both ways, one set toward the sipx box and another toward the sip provider? Any help would be appreciated. Here are some of my settings. I've tried a lot of different combinations in the last couple of days but here is what they are currently. If there are any settings I should or shouldn't have for using OSBC with SipX, please let me know. Thanks. radp1399 Some of my settings: SBC-Application-Mode: B2BUpperReg Mode Interface-Address: sip:*:15060 (sipX uses 5060) Enable-Backdoor-Port yes Enable-Trunk-Port yes Enable-Calea-Port yes Trusted-Domain Accept all calls yes Host-Access-List Trust-all-host yes Local-Domain-Accounts Accept-All-Registration yes B2BUA-Routes sip:20...@my...:*]sip:my.sipxbox.com:*;sip-trunk=true sip:*@my.sipprovider.com:*]sip:my.sipprovider.com:*;sip-trunk=true sip:*@my.othersipprovider.com:*]sip:myother.sipprovider.com:*;sip-trunk=true [sip:*@my.sipxbox.com:*]sip:my.sipxbox.com:*;sip-trunk=true Catch-All-Route blank Outbound-Proxy my.sipprovider.com Upper-Registration Enable-Stateful-Reg yes [sip:*@my.sipprovider.com:*]sip:my.sipprovider.com:*;domain=mysipprovider.com;sip-trunk=true [sip:*@my.othersipprovider.com:*]sip:my.othersipprovider.com:*;domain=my.othersipprovider.com;sip-trunk=true [sip:*@my.sipxbox.com:*]sip:my.sipxbox.com:*;domain=my.sipxbox.com;sip-trunk=true SIP-Trunk-Config <root> <siptrunk trunk-name="my.sipprovider.com" route-set="my.sipprovider.com" sip-domain="my.sipprovider.com" expires="10"> <trunk-accounts> <account user-name="sipproviderusername" auth-user-name="sipproviderusername" auth-password="sipproviderpassword" inbound-route="sip:20...@my..." expires="3600"/> </trunk-accounts> <transient-accounts> <account user-name="sipproviderusername" auth-user-name="sipproviderusername" auth-password="sipproviderpassword" inbound-route="sip:20...@my..." expires="3600"/> </transient-accounts> </siptrunk> </root> |
From: OpenSBC F. <ope...@op...> - 2008-11-12 15:33:18
|
Joegen, I've emailed you the log file. Thanks. radp1399 |
From: OpenSBC F. <ope...@op...> - 2008-11-12 15:05:49
|
I added B2BUA route [sip:100] sip:sipx.ebc.fleurop Unfortunately it didn't help. I will send the logby mail. Thanks for reply. |
From: OpenSBC F. <ope...@op...> - 2008-11-12 14:24:59
|
Your XML file looks straight forward to me. If you can send me a copy of your log, perhaps we can get more info why it did not work. If you can't post it here for security reasons, you can send it to me directly at jo...@op... > {quote:title=radp1399 wrote:}{quote} > > I'm not able to register any sip trunks. I have loaded the following info into SIP-Trunk-Config: > > <root> > <siptrunk trunk-name="xxx.xxx.com" > route-set="xxx.xxx.com" > sip-domain="xxx.xxx.com" > expires="10"> > <trunk-accounts> > <account user-name="xxxx" > auth-user-name="xxxx" > auth-password="xxxx" > inbound-route="sip:20...@si..." > expires="3600"/> > </trunk-accounts> > <transient-accounts> > <account user-name="xxxx" > auth-user-name="xxxx" > auth-password="xxxx" > inbound-route="sip:20...@si..." > expires="3600"/> > </transient-accounts> > </siptrunk> > </root> > > > From the logs it doesn't appear to have even tried to register. The info does show up in OpenSBC.ini. In OpenSBC - General Parameters, I have enabled 'Enable Trunk Port' and restarted. I have tried these and another set of trunks, both of which have been used with asterisk so I know they work, but from looking at the logs it doesn't appear they are even trying to register. Another thing I noticed is that there is a file in the opensbc directory named sip-trunk.conf.xml which only has a default-looking configuration in it. Is this supposed to be filled by what I entered in the SIP-Trunk-config window? Am I missing something? Any help would be appreciated. > > > Thanks. > > > radp1399 > > |
From: OpenSBC F. <ope...@op...> - 2008-11-12 14:22:10
|
You might need to put a b2bua route for sip:10...@si...domain. If that does not work, send a level 5 log. > {quote:title=Vitaly wrote:}{quote} > > I've configured SIP Trunk on OpenSBC. Calls from sipX to PSTN over sip provider are working well. But when I try to call to my external PSTN number (must be routed to sipX number) I got an Internal error after vey long timeout. > > Seems OpenSBC doesn't route calls to sipX. > > Help me pls! > > My configs: > > SBC Mode: B2BOnlyMode > > Interface: sip:*:9000 > > B2BUA Route: [sip:44*] sip:provider.org;sip-trunk=true > > <xml> > <siptrunk trunk-name="provider.org" route-set="provider.org" sip-domain="provider.org" expires="20"> > <trunk-accounts> > <account user-name="44444444444" auth-user-name="44444444444" auth-password="mypass" inboundroute="sip:10...@si...domain" expires="3600" /> > </trunk-accounts> > <transient-accounts> > <account user-name="44444444444" auth-user-name="44444444444" auth-password="mypass" inboundroute="sip:10...@si...domain" /> > </transient-accounts> > </siptrunk> > </xml> > > |
From: OpenSBC F. <ope...@op...> - 2008-11-12 14:07:03
|
I've configured SIP Trunk on OpenSBC. Calls from sipX to PSTN over sip provider are working well. But when I try to call to my external PSTN number (must be routed to sipX number) I got an Internal error after vey long timeout. Seems OpenSBC doesn't route calls to sipX. Help me pls! My configs: SBC Mode: B2BOnlyMode Interface: sip:*:9000 B2BUA Route: [sip:44*] sip:provider.org;sip-trunk=true <xml> <siptrunk trunk-name="provider.org" route-set="provider.org" sip-domain="provider.org" expires="20"> <trunk-accounts> <account user-name="44444444444" auth-user-name="44444444444" auth-password="mypass" inboundroute="sip:10...@si...domain" expires="3600" /> </trunk-accounts> <transient-accounts> <account user-name="44444444444" auth-user-name="44444444444" auth-password="mypass" inboundroute="sip:10...@si...domain" /> </transient-accounts> </siptrunk> </xml> |
From: OpenSBC F. <ope...@op...> - 2008-11-11 23:11:48
|
I'm not able to register any sip trunks. I have loaded the following info into SIP-Trunk-Config: <root> <siptrunk trunk-name="xxx.xxx.com" route-set="xxx.xxx.com" sip-domain="xxx.xxx.com" expires="10"> <trunk-accounts> <account user-name="xxxx" auth-user-name="xxxx" auth-password="xxxx" inbound-route="sip:20...@si..." expires="3600"/> </trunk-accounts> <transient-accounts> <account user-name="xxxx" auth-user-name="xxxx" auth-password="xxxx" inbound-route="sip:20...@si..." expires="3600"/> </transient-accounts> </siptrunk> </root> >From the logs it doesn't appear to have even tried to register. The info does show up in OpenSBC.ini. In OpenSBC - General Parameters, I have enabled 'Enable Trunk Port' and restarted. I have tried these and another set of trunks, both of which have been used with asterisk so I know they work, but from looking at the logs it doesn't appear they are even trying to register. Another thing I noticed is that there is a file in the opensbc directory named sip-trunk.conf.xml which only has a default-looking configuration in it. Is this supposed to be filled by what I entered in the SIP-Trunk-config window? Am I missing something? Any help would be appreciated. Thanks. radp1399 |
From: OpenSBC F. <ope...@op...> - 2008-11-09 23:43:05
|
Hi, Maybe I should have asked if OSBC + Asterisk is the recommended solution if I wanted to do codec transcoding. regards, Miced |
From: Joegen E. B. <joe...@gm...> - 2008-11-09 12:22:43
|
OpenSBC Forum wrote: >> That really depends on your setup and what you mean by "integrated". >> Joegen >> > I meant having Asterisk on the same box just as we can have OpenSER/Asterisk or OSBC/SipXecs. > > > Yes. As long as you use different ports for asterisk and opensbc. Joegen |
From: OpenSBC F. <ope...@op...> - 2008-11-09 08:26:38
|
> {quote:title=Guest wrote:}{quote}OpenSBC Forum wrote: > > That really depends on your setup and what you mean by "integrated". > Joegen I meant having Asterisk on the same box just as we can have OpenSER/Asterisk or OSBC/SipXecs. regards, Miced |
From: Joegen E. B. <joe...@gm...> - 2008-11-09 02:43:27
|
OpenSBC Forum wrote: > Can Asterisk be integrated with OSBC somehow to allow transcoding of Codecs and perhaps IVR? > > That really depends on your setup and what you mean by "integrated". OpenSBC have multiple means to route call to any destination. You first choice would be to use B2BUA routes and point egress calls towards asterisk. You could also set asterisk as an outbound proxy to let outbound calls "always" hit asterisk using Outbound Proxy setting. Joegen |
From: OpenSBC F. <ope...@op...> - 2008-11-09 01:18:14
|
Thanks again for that Can Asterisk be integrated with OSBC somehow to allow transcoding of Codecs and perhaps IVR? |
From: OpenSBC F. <ope...@op...> - 2008-11-09 00:59:42
|
Yes, that's exactly how OSBC is meant to be used. |
From: OpenSBC F. <ope...@op...> - 2008-11-09 00:37:21
|
Thanks for your reply. So If I have a Brekeke SIP Server and Radius Billing in Dallas, can I deploy an OSBC box configured with Upper Registration in New York and one in London for example for handling NAT Traversal and Proxying the Media Stream? Is this sort of how OSBC would be ideally used in the Voip providers network? regards, Miced |
From: OpenSBC F. <ope...@op...> - 2008-11-08 03:45:30
|
Thanks, that worked. |
From: OpenSBC F. <ope...@op...> - 2008-11-08 02:41:02
|
OSBC and Brekeke SIP Server ar every similar in function, although Brekeke does have more features (OSBC only supports Solegy's accounting module, and does not support TCP for signalling; neither can perform codec translation). On the other hand, one might argue that OSBC is more scalable than Brekeke since it is developed in C++ while Brekeke is a Java application. Also, OSBC is open source and free to download, while the Brekeke SIP server costs from $500-$4,000 per instance. |
From: OpenSBC F. <ope...@op...> - 2008-11-08 01:10:40
|
Hello All, I am just getting to understand OpenSBC and wondering how would OpenSBC compare to Brekeke. Could I for example replace it with OpenSBC since they seem to have fairly similar features? Does OpenSBC support Radius? Does OpenSBC do codec transcoding? regards, miced |
From: OpenSBC F. <ope...@op...> - 2008-11-07 09:31:27
|
Hi, yes, you should compile first opesipstack, then you should compile opensbc. before you run configure script for opensipstack, you can check configure options this way: ./configure --help then for compiling you need to run: make bothnoshared . If everything goes without problems, then you can compile opensbc. |
From: Joegen E. B. <joe...@gm...> - 2008-11-07 00:42:45
|
Make sure you build OpenSIPStack first before building OpenSBC. OpenSBC Forum wrote: > Anebi, > > > I installed sipxecs 3.10.2 with the install disk and then group installed 'Development Tools' and 'Development Libraries' with yum. Once that was complete, I unzipped and untarred OpenSBC-1.1.5-RC1-Bundle.tar.gz. Then I ran ./configure and once that was done I ran 'make bothnoshared'. The ran for a bit when I got the errors I referred to above. > > > Thanks. > > > radp1399 > > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > ------------------------------------------------------------------------ > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.175 / Virus Database: 270.9.0/1771 - Release Date: 11/6/2008 7:58 AM > > |
From: OpenSBC F. <ope...@op...> - 2008-11-06 19:05:12
|
Anebi, I installed sipxecs 3.10.2 with the install disk and then group installed 'Development Tools' and 'Development Libraries' with yum. Once that was complete, I unzipped and untarred OpenSBC-1.1.5-RC1-Bundle.tar.gz. Then I ran ./configure and once that was done I ran 'make bothnoshared'. The ran for a bit when I got the errors I referred to above. Thanks. radp1399 |