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37510 discussions

[Serusers] Alternate port for rewritehost
by Stephen Helfen 24 Aug '06

24 Aug '06

All,I am receiving SIP traffic on a non-5060 port but need to change the port of the rewritehost to 5060 as the carrier is dropping my INVITE messages with the non-5060 port. Any suggestions?Thanks,Steve _________________________________________________________________ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://get.live.com/toolbar/overview

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[Users] Please help for guiding serweb installation in openser server
by Ferianto siregar 24 Aug '06

24 Aug '06

Dear all, Thank you very much to all of you for help me to answer my question about serweb before. So, based on your suggestion, I tried to install serweb 0.9.3 for my openser server. But, when I tried to install it, I am confused. I don`t understand about the installation note that I have got from untar the serweb 0.9.3 packet. These are some questions that I need to ask: 1. Where should we untar the serweb 0.9.3 packet? Should we untar it in root or anywhere we like? Because when we untar it, there will be a serweb folder in the location where we untar it (for example, when we untar the packet in root there will be a serweb directory in root) . This serweb folder also contain html directory and phplib directory. As I know, There are also html directory in /root/var/www. So, What directory should I use? Should I delete the html directory and phplib directory in /root/var/www/? 2. Is there any relationship between the html directory in /root/var/www and html directory from /root/serweb? Which one should I use? 3. In the installation note I got this message : "install it, unpack the distibution on your webserver so that the directory html is in document root and phplib is not accessible via web" Can anybody help me to explain what this message means? and give me an example? 4. If I can make a wish for all of you, I do hope anybody can share the serweb configuration that she/he has configured to me. So, it can become the guidelines for me. That `s all the question that I got when I tried to install the serweb. I am confused. I do hope anybody can help me because I am new in this system. Thank you very much for your help. Thanks Regards, Ferianto --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

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[Users] NAT (again)
by Mark Kent 24 Aug '06

24 Aug '06

I'm testing my rtpproxy config with a linksys phone and a grandstream phone, either not NAT'ed, or behind a cisco (full cone), or behind a Netgear (symmetric). Things generally work, but I have a problem with the grandstream behind the symmetric nat. This is the register: U NAT-IP:3284 -> PROXY-IP:5060 REGISTER sip:sip.example.com SIP/2.0 Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK0103d0ff79cc9c02 From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=743335557afcade4 To: <sip:frick@sip.example.com;user=phone> Contact: <sip:frick@NAT-IP:3280;user=phone> Authorization: Digest username="frick", realm="sip.example.com", algorithm=MD5, uri="sip:sip.example.com", nonce="foo", response="bar" Call-ID: 11f6defb0857db17(a)192.168.0.100 CSeq: 116 REGISTER Expires: 60 User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 This is the resulting contact: ...Record(0x2864b680).. domain: 'location' aor : 'frick(a)sip.example.com' ~~~Contact(0x2864d4e0)~~~ domain : 'location' aor : 'frick(a)sip.example.com' Contact : 'sip:frick@NAT-IP:3280;user=phone' Expires : 34 q : Call-ID : '11f6defb0857db17(a)192.168.0.100' CSeq : 119 User-Agent: 'Grandstream BT100 1.0.6.7' received : 'sip:NAT-IP:3284' Path : '' State : CS_SYNC Flags : 1 Sock : PROXY-IP:5060 (0x8122f50) Methods : 5695 next : 0x0 prev : 0x0 ~~~/Contact~~~~ .../Record.. Now, what I don't understand is the two ports used above: 3280 and 3284. It looks like the phone thinks it is being natted to port 3280, but it's really getting port 3284 When I make a call I get U NAT-IP:3284 -> PROXY-IP:5060 INVITE sip:16509415674@sip.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK4a4c039fbcb5bef8 From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=2dc2fc3c94e93951 To: <sip:16509415674@sip.example.com;user=phone> Contact: <sip:frick@NAT-IP:3280;user=phone> Supported: replaces Call-ID: 1f04f9c6ec474f70(a)192.168.0.100 CSeq: 26381 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 419 v=0 o=frick 8000 8002 IN IP4 NAT-IP s=SIP Call c=IN IP4 NAT-IP t=0 0 m=audio 3295 RTP/AVP 0 8 4 18 2 15 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 and my side sends back the Proxy Authentication Required, but it sends it to port 3280. Now, is that based on the Via: header or the Contact: header, or (unlikely) the stored contact info from the REGISTER? The phone acts like it never got it (and I'm guessing it didn't) and sends the INVITE again without any auth data. The thing that bugs me is that this is an outbound call, from the phone out through the NAT box, I figured that would be doable. The Linksys942 works, so I'm hoping this is a Grandstream bug, or something I can fix in my openser config. Thanks, -mark

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[Users] Limiting Call Volume and CPS
by Brian Jones 24 Aug '06

24 Aug '06

Hi, I was wondering if anyone had any suggestions on how I could use OpenSER to limit call volume and call per second on a peer basis. Thanks! Brian.

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[Users] Weird forwarding with Route header
by Jeremie Le Hen 24 Aug '06

24 Aug '06

Hi list, my proxy (yy.yy.yy.yy) receives a message like this: BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> My ruleset is something like this (with the REGISTER logic pulled out) : %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; force_rport(); if (!method == "REGISTER") { record_route(); }; if (loose_route()) { route(1); }; loopkup("location"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% I would expect the packet to leave my router, back to xx.xx.xx.xx : BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:yy.yy.yy.yy;lr;ftag=2873864> Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP yy.yy.yy.yy;branch=zu83fdhDB8d.67235676.0 Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> But the outgoing packet is : BYE sip:xx.xx.xx.xx;ftag=1056999653;lr=on SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP yy.yy.yy.yy;branch=z9hG4bK07cb.02005244.0 Via: SIP/2.0/UDP xx.xx.xx.xx;rport=5060;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Do you have any idea about what is going on ? Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] dropping a 200
by Simon Morvan 24 Aug '06

24 Aug '06

Just to be sure : there's no way to call drop() in an onreply_route block when it's not a provisionnal reply ? -- Simon.

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[Users] how to correctly handle NAT on inbound call from another SIP-proxy?
by Gerd Feiner 24 Aug '06

24 Aug '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi list, we have some trouble handling NAT correctly when receiving a call from another SIP-proxy. the mediaproxy-module does not seem to rewrite SDP when it finds another proxy's record-route header, though it invokes mediaproxy and sends uncorrected SDP to the other SIP-server resulting in one-way-audio as the provided c=-field includes the RFC1918-address of the client. when receiving calls through our Cisco AS5350 everything is fine ('cause it does not add the record-route). the addition of the header cannot be circumvented bacause the other provider wants to bill calls (obviously) and therefor needs to be in the SIP-dialog. is there a neat solution to this problem or do we have to get a session-border-controller or some B2BUA-config for handling the received calls? and by the way, how do i distinguish external INVITEs destined for one of our clients (uri==myself) from the INVITEs being sent out to the callee upon an external INVITE? if there was a way we could invoke mediaproxy from there? how is the recommended setup? brgds, g.feiner -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE7aVOqvDCGdeEb8oRArDjAJ9ssHEHBAIGmSB34oHQxuuT0ke15wCgmKvB chyomyJgqU9NVMep7QV1nFI= =r5Xb -----END PGP SIGNATURE-----

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[Users] Q:ABOUT:MODULE:auth_radius
by Alejandro Sanchez 24 Aug '06

24 Aug '06

Hello. openser 1.1 radiusclient-ng 5 freeradius 1.1.1 I need to enable the modules *_radius for my openser instalation a follow the guide for enable radius then when i try to start openser i get the following error. 0(0) ERROR: auth_radius: can't get code for the Digest-Response attribute Anyone knows what is the cause for this error? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

Hi, Is there a way to coexist multiple domains in Ser, distinguishing users even if they have the same username in subscriber table but different domains? I'll explain better, I have subscriber table with: | username | domain | e-mail | ------------------------------------------------ | user_A | domain_A | user_A@domain_A | ------------------------------------------------ | user_A | domain_B | user_A@domain_B | and in domain table there are both domains domain_A and domain_B. But when someone calls user_A, (with or without writing their domain), both phones ring (if both users have their phones registered), when I want only one to ring because they are different users!! Is there a way in Ser to do this right? Regards, Ricardo.

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Re: [Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

It's true Asterisk can use different contexts, but that doesn't solve my problem because I use Ser + Asterisk and Ser is used for routing SIP messages between SIP UAs, not Asterisk. In my architecture, Asterisk is only used for advanced services like voicemail or call conferencing and for gateway, not to route messages between UAs. Ricardo. openser(a)comcast.net wrote: > The way to do that is intergrate Ser with Asterisk and in Asterisk you > can build context and keep the same numbers for different context like > a pbx. > > > -------------- Original message -------------- > From: Ricardo Carvalho <rcarvalho(a)iric.up.pt> > > > Hi, > > > > Is there a way to coexist multiple domains in Ser, > distinguishing users > > even if they have the same username in subscriber table but > different > > domains? > > > > I'll explain better, I have subscriber table with: > > > > | username | domain | e-mail | > > ------------------------------------------------ > > | user_A | domain_A | user_A@domain_A | > > ------------------------------------------------ > > | user_A | domain_B | user_A@domain_B | > > > > and in domain table there are both domains domain_A and domain_B. > > > > But when someone calls user_A, (with or without writing their > domain), both > > phones ring (if both users have their phones > > registered), when I w! ant onl y one to ring because they are > different users!! > > Is there a way in Ser to do this right? > > > > Regards, > > > > Ricardo. > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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