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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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[Users] 1.1.x/permissions.html is not up to date
by Juha Heinanen 01 Sep '06

01 Sep '06

for some reason web page http://www.openser.org/docs/modules/1.1.x/permissions.html is not up to date. for example, 1.3.12. tag_col (string) 1.3.13. peer_tag_avp (string) are missing. they do exist in rel_1_1_0 source. -- juha

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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] SIP Contact header field issue incase of NAT
by Abdul Qadir 01 Sep '06

01 Sep '06

Hi All, I have problem connecting Nokia Phone with SER. The problem is When SER receives the registration request from the Nokia phone. The registration request contain the phone's private IP (192.xxx.xx.xx), since we use wifi for connection. Our proxy detects that since this is a private IP handles the NAT translation and determines the phone public IP address (e.g., 202.xxx.xx.xx, port xx). Our SIP proxy informs the phone that registration is successful but the contact information is changed from userid(a)192.xxx.xxx.xx to userid(a)202.xxx.xxx.xx, port xx. On recieving this changed contact info the Phone rejects 200 Ok. We tried to contact nokia people and they said that its part of RFC3261 to reject such contacts. After this I have also gone through RFC 3261 section "10.2.4" and it seems that phone is doing rite. Can any body tell me why SER is changing the contact address in case of NAT. It would be higly appreciated if you can give me some reference from some RFC or other document which I can show to nokia people. Best Regards, Abdul Qadir --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

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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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[Serusers] mediaproxy
by Javier Ramirez 01 Sep '06

01 Sep '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

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[Users] openser routing lookup question
by Dave Logan 01 Sep '06

01 Sep '06

Hey all, I have what may or may not be a simple situation. I've read through dozens of forum entries, dozens more list mails, and a handful of "tutorials" for openser, and I'm still missing something I think. I'm not afraid of the documentation, but neither the "core cookbook" nor the modules documentation had enough detail that I could catch on to what's going on. Let me lay out my scenario, and maybe somebody can point to a particular document or solution I can follow up on. I have multiple asterisk boxes: four as media gateways with TDM trunks, one feature server (VM), and three registrars. So needless to say, I have a lot of dialplan in there just for calls that stay in my network. What I'd like is to introduce a single openser box in there, and for any call that any of those asterisk machines gets, it sends it to openser, which then just calls a simple script (which I've already more or less written) to know what to do. So for example, I own the 555-1000(-1099) DID block DID 555-1000 is registered to machine1 DID 555-1001 is registered to machine2 In openser, I'm envisioning a call like such (php style notation): my $number = 5551000 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551000@machine1 or my $number = 5551001 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551001@machine2 Then I can just route it in the normal openser way. So there, my lookupdestination script knows all the numbers in my central database and where they're located, and on an unknown number assumes it's outside our network so just sends it to the gateway as is (5552000@gw1) or so. Thanks for any pointers at all, dave -- Dave Logan http://www.digitalcoven.com/ "Do let's pretend that I'm a hungry hyena, and you're a bone!" - Alice

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