August 2003 - sr-users - kamailio.org

Re: [Serusers] rewrite & ACK forwarding problem
by lgfausak＠august.net 30 Apr '04

30 Apr '04

I have also had problems with getting the ACK back. I don't completely understand your configuration, you must allow for packets going both directions, right? Here is my config : route { # check to see if the message has been around too long # probably means that it is looping # if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; # # make sure the length of the message isn't too long! # if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; # # do the loose-routing thing, this is important! # if(loose_route()) { log(1,"doing top loose route"); t_relay(); break; }; # this is where I was dropping the ACKS. # I was simply dropping these, but they must be relayed # because they can be ACKs if(!(uri==myself)) { if(!t_relay()) { sl_reply_error(); break; }; break; }; This gets the ACKs through for me. By the way, I have this configured with Cisco ATAs, version 2.16. ---greg > >I have the same problem and posed it to the group yesterday ([Serusers] >Ignored 200 OK message.) So far the only workaround that I have found is to >use the rules in my gateway to rewrite the dialed digits before sending them >to the PSTN PRI, thus leaving the origianl URI intact for SIP >communications. > >One person told me that this is a bug in the Cisco ATA, but it happens on my >IPDialog phones also. It seems to me that the INVITE is being processed by >the SER dial rules and is rewritten, but the ACK is not. > >Sean >_______________________________________________ > >Sean Robertson > >NETXUSA >p. 800-289-6389 >f. 864-233-4344 "Ask me about Voice over IP." >http://www.netxusa.com/ > >----- Original Message ----- >From: "Alexander Mayrhofer" <axelm(a)nic.at> >To: <serusers(a)lists.iptel.org> >Sent: Friday, June 27, 2003 12:15 PM >Subject: [Serusers] rewrite & ACK forwarding problem > > >> >> Hi, >> >> we're running SER together with a PSTN Gateway. Before a call get's >> forwarded to the gateway, we are rewriting the request URI to make >> rewriting on the GW as simple as possible: >> >> route { >> ... >> strip(3); # +43xxx -> xxx >> prefix("0"); # xxx -> 0xxx >> rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway >> route(1); >> break; >> ... >> >> SIP call flow looks like (record route enabled): >> >> (1) phone -> SER >> INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0 >> >> (2) SER -> phone >> SIP/2.0 100 trying -- your call is important to us >> >> (3) SER -> GW >> INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> (4) GW -> SER >> SIP/2.0 100 Trying >> >> (5) GW -> SER >> SIP/2.0 183 Session Progress >> >> (6) SER -> phone >> SIP/2.0 183 Session Progress >> >> (7) GW -> SER >> SIP/2.0 180 Ringing >> >> (8) SER -> phone >> SIP/2.0 180 Ringing >> >> (9) GW -> SER >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxxx@xx.xx.xx.xx:5060> >> >> (10) SER -> phone >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxx@xx.xx.xx.xx:5060> >> >> [ call established, we can talk, but ... ] >> >> (11) phone -> SER >> ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> --> Here starts the problem. That ACK (11) never gets forwarded to the >> Gateway, so after a few seconds, the GW starts over at (9). Those three >> packets (9-11) repeat a few times until GW runs into a timeout and drops >> the call. >> >> I have the impression that SER can't match the packet to the previous >> requests because of the rewritten URI. Is that correct? >> >> The only output at debug level 3 is: >> >> Warning: sl_send_reply: I won't send a reply for ACK!! >> >> Is that a routing goof somewhere in our scripts or is that a more >> generic problem? Is the problem that the warning indicates somehow >> related to the fact that the ACK is not being forwarded? >> >> Help appreciated. >> >> cheers >> >> axelm >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

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RE: [Serusers] Latest STABLE build without using CVS
by Jiri Kuthan 20 Jan '04

20 Jan '04

At 07:39 PM 8/14/2003, Chad Brown wrote: >Perfect, > >Let me ask 2 quick follow-on questions... > >1. Can I go back to http://www.iptel.org/ser/tarball/ser_8_11_stable.tgz >to get the latest patched and STABLE builds? That's the latest, most stable source, you need to compile it myself. There will be a new complete distribution by end of this month -- we are now waiting to run SER through the upcoming SIPIT to release it. >2. What location / version of the modules should I use when running >builds for this location? (Serweb, mysql, jabber, etc) All SER modules are included there. uptodate SERWEB is now in the http://www.iptel.org/ser/tarball/ directory too. -jiri

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[Serusers] Xten and SER problem
by Jesus Rodriguez 17 Nov '03

17 Nov '03

Hello, I'm trying to register an Xten (1016 build) to SER and i always get a "401" response. Registering an ATA186 to the same SER with the same user and pass works with no problem (logs below Xten). We have tried SIPPS and registers ok too. Xten send the REGISTER, SER answers with a WWW-Authenticate challenge but Xten doesn't send anything, just sends a new REGISTER after few seconds: (c)2003 Xten Networks Inc. All rights reserved. Private build: 1016 SIP: 192.168.1.203:5060 RTP: 192.168.1.203:8000 NAT: 192.168.1.203 PXY#0: 192.168.1.2:5060 SEND >> 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2> Contact: "JesusR" <sip:1000@192.168.1.203:5060> Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER Expires: 500 User-Agent: X-Lite build 1016 Content-Length: 0 RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562000000000dded3e97ed7f89d3722ff12d3704a884",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13588 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562000000000dded3e97ed7f89d3722ff12d3704a884",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13589 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" SEND >> 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2> Contact: "JesusR" <sip:1000@192.168.1.203:5060> Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31485 REGISTER Expires: 500 User-Agent: X-Lite build 1016 Content-Length: 0 RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31485 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562200000000e06a238bd1858769f2e46b11d8d837f1",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13588 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" The same action works ok with ATA186: [0:0] Tx Msg to 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone> Call-ID: 2622180140(a)192.168.1.202 CSeq: 1 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;expires=60 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Content-Length: 0 [0]Rx Msg from 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone>;tag=d907c037823644515dfe0ede38ca9976.9cf4 Call-ID: 2622180140(a)192.168.1.202 CSeq: 1 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c579a00000000aa6fd2432e429f54e3 7e75981218834b",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13590 req_src_ip=192.16 8.1.202 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" [0]Reg Resp 401; Unauthorized [0]REGISTER Retry 0 [0:0] Tx Msg to 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone> Call-ID: 2622180140(a)192.168.1.202 CSeq: 2 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;expires=60 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Authorization: Digest username="4070",realm="voztelecom.net",nonce="3f1c579a0000 0000aa6fd2432e429f54e37e75981218834b",uri="sip:192.168.1.2",response="fe07b8be83 0305cca7a28bf35c112270" Content-Length: 0 [0]Rx Msg from 192.168.1.2:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone>;tag=d907c037823644515dfe0ede38ca9976.9cf4 Call-ID: 2622180140(a)192.168.1.202 CSeq: 2 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;q=0.00;expires=6 0 Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13589 req_src_ip=192.16 8.1.202 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" Thanks in advance. Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

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[Serusers] FCP module in SER
by Kevin Chu 25 Sep '03

25 Sep '03

Hi, It's mentioned in the release notes of SER 0.8.11 that there is a a new experimental features called fcp. Is that module the implementation of fcp-client mentioned in this discussion? http://lists.iptel.org/pipermail/serusers/2003-January/000111.html I can't find the fcp module from anywhere. May I know where I can find it and the latest status? Thanks, Kevin

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[Serusers] Questions about address SIP numeric ans gateway sip-PSTN???
by Hassan CHOUMAR 15 Sep '03

15 Sep '03

Hello all, I installed serweb, it works well, but I have a problem at the level of the application: when I want to make a subscription with an address numeric SIP for example: 9991(a)domaine.com, it not accepts it , it sends me a message (user name does not follow suggested convantions), Another question, if I want to try to make communications between a softphone SIP and an analogical telephone through a cisco router, it is enough to go the commands this below into the cisco router, and the others into the server SER: In the router: Dial-peer voice 999 voip Destination-pattern 555999. ** Associate the number arranges 555-9990 to 9999 with our SIP server Session protocol sipv2 ** Set this dial-peer to uses(wears out) SIP instead of Cisco protocols Session target sip-server ** Send the call to our SIP server. See SIP-UA below Codec g711ulaw ** Set the default codec to 711-Ulaw (common codec between customers) ! Sip-ua Sip-server ipv4:192.168.0.1 In the fileser.cfg : if (uri=~"^sip:9[0-9]*@mydomain.com") { ## This assumes that the caller is log("Forwarding to PSTN\n"); ## registered in our realm t_relay_to( "192.168.0.2", "5060"); ## Our Cisco router break; }; thanks you for advance hassan ------------------------------------------------- This mail sent through IMP: http://horde.org/imp/

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[Serusers] Grandstream Phones
by Gavin Bensom 02 Sep '03

02 Sep '03

Hello, Has anyone used the Grandstream BudgeTone-100 phones with SER? When I try to transfer a call the call gets disconnected. The grandstream documentation shows the transfer protocol here: http://www.grandstream.com/user_manuals/budgetone100.pdf [pg 15,16 of the .pdf] Any ideas what is happening? Is the Grandstream protocol compliant with what SER expects for transferring calls? Also, does SER support the 484 Incomplete address response? The BudgeTone phone can be set to send an INVITE after every button press, and will continue to do so if the server response with incomplete address instead of 404 not found. Finally, if users passwords are set to (null) will a phone that registers with a username but no password successfully register? [I haven't tried it yet]. Thanks, G. --------------------------------- Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo.

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Re: [Serusers] serweb configuration on alternate port
by Samy Touati 01 Sep '03

01 Sep '03

Here is the confirmation email I receive. I do add the port 5001 when I paste the link into the browser. Samy. ----- Original Message ----- From: Jiri Kuthan <jiri(a)iptel.org> Date: Tuesday, August 26, 2003 10:37 am Subject: Re: [Serusers] serweb configuration on alternate port > Can you forward us the confirmation email? -jiri > > At 04:38 PM 8/26/2003, Samy Touati wrote: > >Hi, > > > >I installed ser along with mysql and serweb. > >I'm able to login in admin mode using serweb. > >I tried to create new users using the subscribe > >button, the process goes smootly, and I do get an email > >asking me to confirm. > >When I do click on the onfirmation link, the browser > >tries to access the confirmation page, but I never > >see it, and I never see any error (page not found..) it just goes > >balnk forever. > >In fact I can never see any link generated by php > >and sent via email (password retrieval, account confirmation). > >I'm running http on port 5001, but I'm taking care of adding > >the port when accessing the pages, but to no effect. > > > >Is anyone using serweb, is there something I should pay attention > to > >if changing the default port number of apache ? > > > >Thanks. > > > >Samy. > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ > >

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[Serusers] Contact Table
by giAndrea 01 Sep '03

01 Sep '03

Hi I have some questions about Contact Table: What is "q" field? And "callid"? How can I add rows on server_monitorig table? Thanks. Andrea ---------------------------------- Legge di Clarke sulle idee rivoluzionarie Ogni idea rivoluzionaria - in campo scientifico, artistico o altro provoca tre stadi di reazione, riassumibili nelle seguenti frasi: 1. "E' impossibile; non farmi perdere tempo". 2. "E' possibile, ma non val la pena di farlo". 3. "L'ho sempre detto, io, che era un'ottima idea".

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[Serusers] Interoperability
by Annie Sasidar 01 Sep '03

01 Sep '03

Hi, The UA application on a mobile device will be developed using WindowsCE.NET 4.2 and the SIP Proxy and Registrar Server will be a Linux box. Is there any interoperability issues that i need to know about. Experts please advice. Also, any tips on how to go about the implementation of a Stateless Proxy and Registrar Server on a Linux box will be of great help. I did download the ser free server from the website. Could anybody guide me as to where to start working with the package i obtained. I am new to this so please pardon my ignorance. I am keen on learning and improving. Please guide. Thanks.

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[Serusers] On the diversity of instant message protocols
by kaiduan xie 01 Sep '03

01 Sep '03

Hi, all, Now there are so many different instant message services and clients, for instance, AOL, MSN, YAHOO messager, and so on. I have a question on the diversity of these clients and the protocols behind. As we know, SIP has gain more and more momentuams. Will SIP replace all the other instant message protocols? If yes, how long will this process take? Besides, can anyone tell me where can I find the protocol used in MSN messager (not Windows messager), YAHOO messager and AOL messager, repectively? Thanks a lot for help. kaiduan ______________________________________________________________________ Post your free ad now! http://personals.yahoo.ca

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