sr-users August 2006

sr-users@lists.kamailio.org

184 participants
360 discussions

[Users] VIA Parameter?
by Ricardo Martinez 01 Sep '06

01 Sep '06

Hello list. Sorry to ask this question here, but i can't find the answer in the RFC3261 Is this "VIA" permited according to the RFC? Via: SIP/2.0/UDP 69.20.6.205:15060, SIP/2.0/UDP 69.1.229.69:4081;branch=z9hG4bK-58e9ccfcef0f4c99915b8007cb9713e1-1156793814 It seems to be two VIA's in ONE line... Thanks Regards, Ricardo.-

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[Users] SIP Redirection in Openser
by Padmaja RV 01 Sep '06

01 Sep '06

Hi all, I have tried implementing SIP redirection as mentioned in the example script below: route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # rewrite current URI, which is always part of destination ser rewriteuri("sip:parallel@iptel.org:9"); # append one more URI to the destination ser append_branch("sip:redirect@iptel.org:9"); # redirect now sl_send_reply("300", "Redirect"); } But that does redirection only for that account as the account is included in the code. I need the code to be able to redirect any call. Can you tell me how I can generalize the code so that redirection works for any account I choose to redirect? Thanks in advance, Padmaja

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[Users] Authentication in Trunk-Mode
by Martin Klisch 01 Sep '06

01 Sep '06

Hey, are there working ways for authentication with sip trunks? i've read cisco gateways (PGW) and cisco call manager can do this. anyone knows if openser+asterisk support it too?

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[Users] About 'location' table
by Michel Zenone 01 Sep '06

01 Sep '06

Hi! I have registered, thanks to openserctl, an user :'sip:testeur@192.168.0.102'. But when I tyr to reach him, the transaction isn't considered as local processed one. In location table there is: +username + domain + contact + |testeur | | sip:testeur@192.168.0.120:5080 | +---------+--------+--------------------------------+ -192.168.0.120 is the IP address of the user's phone. -192.168.0.102 is the openser IP address. The openser.cfg is as the initial openser.cfg. Thanks, Michel

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[Users] OpenSER AstLinux build posted for testing
by Kristian Kielhofner 01 Sep '06

01 Sep '06

Hello OpenSer Users, It seems that many of you are also Asterisk users or at least subscribe to asterisk-users. Some of you might recognize AstLinux... Just in case you don't: http://www.astlinux.org ---and--- http://www.sf.net/projects/astlinux [in transition] Anyways, I have long hoped to build a version of AstLinux that includes OpenSer. Basically, what I have done is taken astlinux-trunk, add openser, add rtpproxy, de-select the Asterisk packages, and build! What I got was 11mb (gzipped) disk images ready to be written to CF cards/hard disks. As usual, I have created two images: *net4801.img.gz is for the Soekris net4801 SBC *i586.img.gz is for just about any i586 and higher machine There is also an ISO image ready to boot on or install to most i586 or higher machines. You guessed it, it is the one with .iso in the filename. It includes both the net4801 and i586 images. These are 48mb gzipped disk images, with GRUB bootloader and ext2 root filesystem. If you want to write them to disk, try this: gunzip [image.gz] dd if=[output file.img] of=/dev/sdb # <--- VERIFY THIS DEVICE Where /dev/sdb is where your USB CF flash adapter is. If you want access to the ext2 filesystem, try this: gunzip [image.gz] mkdir /mnt/image mount -t ext2 -o loop,offset=32256 [image.gz] /mnt/image If you are curious about how these images were created, take a look at SVN on the astlinux project page. The OpenSer version used for these builds was openser-1.1.0-notls. I built every module that did not depend on an external library that I don't have yet (sorry, no mysql, jabber, or radius yet). I need help!!! For the time being, I don't have mysql. Also, because AstLinux is targeted for compact flash, mysql isn't a good idea anyways. I was not able to get a working configuration in the time alloted using DBTEXT. Does anyone have a sample config that I can put in this build? As of now, openser, rtpproxy, openser modules, and all of the astlinux-style init scripts are done. All I need is a good sample config file to make it "plug and play". I also need testing! My current to-do list: - add mysql (and the openser modules, of course) - add radius support (freeradius?) - jabber/xmpp (depending on size, complexity, etc) - fix astshape (iproute2-based traffic shaper) RTP TOS values to be the same that rtpproxy uses, and standardize this across all astlinux builds - COME UP WITH A NAME. I have no idea what to call this thing. Any suggestions? SERlinux (or something like it) is too obvious and boring! You can grab the goodies here: http://www.krisk.org/openser/ Regardless, I hope that someone finds this interesting or useful. Let me know what you think! -- Kristian Kielhofner

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[Users] icmp port unreachable
by Helge Waastad 01 Sep '06

01 Sep '06

Hi, (just a little bit on the side-bar) I've had some discussion with the developer of twinkle (which is, BTW, a very nice linux softphone) regarding handeling icmp port unreachable and switching to a lower priority SIP router (by DNS SRV). It seems like the dist I'm using (Dapper, kernel 2.6.15-26) will not easily allow the applications to receive the icmp on an unconnected udp socket, hence failover will not work. However on an 2.6.11.4-21.11 kernel it works. Does anyone know if there are any kernel opts which should be added for having this kind of support? Just my thoughts this morning......

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[Users] Openser + Asterisk for P2P calls
by Lucio Maciel 01 Sep '06

01 Sep '06

Hi, Hi, im using an Openser integrated with Asterisk to route PSTN calls, this is working great, but i also want to route P2P calls to Asterisk like this UA1 -> Openser -> Asterisk (calls DIAL(SIP/xxxxx@openser) -> Openser -> UA2, to use its mediaproxy, and account all calls. When UA1 calls UA2, everything goes fine, UA2 receive the INVITE and sends the OK to UA1, the only thing is that UA2 waits for an ACK from the OK that never comes. Have someone tried to do this? How can i configure openser and asterisk for this? Thanks Lucio Maciel

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[Users] IP authentication
by Brian Fertig 01 Sep '06

01 Sep '06

greetings, I have digest and basic auth setup. I am trying to figure out how to setup IP auth. I have read a lot of documentation but I can't seem to find anything to help me understand how to accomplish this. Can someone point me in a direction to some information on how to set this up? Brian

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[Serusers] extra things confusing auth_radius?
by Natambu Obleton 01 Sep '06

01 Sep '06

Ok. I have two clients setup. One is X-Lite and the other is a polycom 601. I am doing radius authentication, but for some reason it doesn't work with the polycom 601 phone. In the debug the only thing that is different about them is something in the uri. Aug 30 09:16:51 laplata ser[25968]: DEBUG: get_hdr_field: <To> [72]; uri=[sip:311@voip.fasttrackcomm.net] Works, but.. Aug 30 09:17:53 laplata ser[25970]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:311@voip.fasttrackcomm.net;user=phone] <mailto:%5bsip:311@voip.fasttrackcomm.net;user=phone%5d> Doesn't. Like it doesn't even send an authentication request to my radius server. I think it is the ';user=phone' part that is confusing it? So what I want to do is remove that part from the URI as a test. I just added a subst to remove it, but still isn't working with radius. It acts like the realm is no longer local. Here is my config file: http://www.natambu.com/serconf.html Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

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[Serusers] Bandwidth parameter modification (b:) from ser.cfg
by "M. Victoria Peláez" 01 Sep '06

01 Sep '06

Hi! I am a beginner with SIP/SER systems. I have been having a look into ser.cfg, in order to modify some SDP parameters, for instance, fix_nated_sdp to change connection IP, and I have found enough information to do it. But, now, I am traying to change b parameter, but cannot find where to do it (Which module manage it?). I am also looking for a document with a description of all bandwidth types (CT, AS). Could you tell me something about this stuff, please? Thank you very much in advance for your help. I am very sorry if this is a very silly question! Best regards, Victoria

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sr-users August 2006