October 2003 - sr-users

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 >

4 5

[Serusers] If anyone can comment of asterisk problem
by John Foster 30 Jan '04

30 Jan '04

Hi List, i tried on asterisk list as well, but any of you can give any idea abt SIP authenticatin problem with Asterisk and SIP UA (i tried SIPPS and X-Lite) I could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params [12321] type=friend username=12321 host=dynamic secret=ccarta context=default mailbox=1234,2345 ; Mailbox for message waiting indicator [77777] type=friend username=77777 host=dynamic secret=atracc context=default mailbox=1234,2345 m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work. Sip debug messages are pasted below. Best Regards, JF Sip read: REGISTER sip:192.168.100.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71> Call-ID: 990209125-415b5d61@990209122-415b5d5e Contact: ccarta <sip:12321@192.168.100.66:5062>;expires=600;q=0.500 Expires: 600 CSeq: 1 REGISTER Content-Length: 0 User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 10 headers, 0 lines Using latest request as basis request Sending to 192.168.100.66 : 5062 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Content-Length: 0 to 192.168.100.66:5062 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d" Content-Length: 0 to 192.168.100.66:5062 Sip read: REGISTER sip:192.168.100.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71> Call-ID: 990209125-415b5d61@990209122-415b5d5e Contact: ccarta <sip:12321@192.168.100.66:5062>;expires=600;q=0.500 Expires: 600 CSeq: 2 REGISTER Content-Length: 0 Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="00000001",response="b8b1d7fc53eff354dfc31dfa3f800749" User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 11 headers, 0 lines Using latest request as basis request Sending to 192.168.100.66 : 5062 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Content-Length: 0 to 192.168.100.66:5062 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 600 Contact: <sip:12321@192.168.100.71>;expires=600 Date: Tue, 14 Oct 2003 13:46:14 GMT Content-Length: 0 to 192.168.100.66:5062 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:12321@192.168.100.66:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3b From: "asterisk" <sip:asterisk@192.168.100.71>;tag=as3f6e8c0e To: <sip:12321@192.168.100.66:5062> Contact: <sip:asterisk@192.168.100.71> Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1(a)192.168.100.71 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.100.66:5062 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.71;branch=z9hG4bK3ecb7a3b From: "asterisk" <sip:asterisk@192.168.100.71>;tag=as3f6e8c0e To: <sip:12321@192.168.100.66:5062>;tag=3b302259 Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1(a)192.168.100.71 CSeq: 102 NOTIFY User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 Content-Length: 0 8 headers, 0 lines --------------------------------- Do you Yahoo!? The New Yahoo! Shopping - with improved product search

2 2

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

3 2

[Serusers] Radius accounting - radius_log_flag?
by Steven R. Bunin 16 Jan '04

16 Jan '04

Hi, I have radius authentication working thanks to Jan's help but I am still not receiving any accounting messages to my radius server. I found a mention of setting radius_log_flag but ser 0.9.11 tells me it isn't found in the "acc" module. How can I set invite and goodbye messages (or any others for that matter) to send information to my radius server for accounting purposes. I have read at leat 90% of the old messages but have yet to find an answer. Thanks in advance, Steve

3 4

[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 -------------------------------

5 6

[Serusers] how to install ser and configure it with voicemail
by kelvin 06 Nov '03

06 Nov '03

I have install ser-0.8.11-0.i386.rpm, mysql and serweb. My sip phone are all window messenger. When someone contact a user who is not on-line, you get this message:- phone(a)192.168.1.3 has declined your request to have a voice conversation. When I login 'phone' to serwweb as a subscriber, under missed calls is always empty & message store too. How to make it ring first and after a period go to voicemail if phone(a)192.168.1.3 is not online? I went through the document but got confuse. Do you have a example ser.cfg script that works. 2. Through the web documents there are 2 vm softwares sems-0.1.0-0.i386.rpm & ortp-0.6.2.tar.gz. which is the right one? 3. if vm is required must you reinstall ser? If yes, what to change in the makefile? Help! Thanks!

3 2

[Serusers] RADIUS authentication: draft-sterman-aaa-sip
by Jiri Kuthan 03 Nov '03

03 Nov '03

Unfortunately, there is now no standard for use of RADIUS along with SIP. SER users leveraging the combination of these two technologies are left with implementation of expired internet drafts. There are some chances that the IETF community revitalizes the document. Thus, I would appreciate hearing if any of the active RADIUS/SIP/SER users have had any issues with the RADIUS authentication in SER, which is based on draft-sterman-aaa-sip. Thanks, -jiri -- Jiri Kuthan http://iptel.org/~jiri/

4 6

[Serusers] New release of seradmin v .03e
by Xten India 02 Nov '03

02 Nov '03

Hai, We Xten released a newer version (seradmin v .03e) of SERAdmin. Xten is building this for the ser community and we would very much like your feedback so that we can build you a better product The Source code is available at http://developer.berlios.de/projects/seradmin SERAdmin is a GUI interface between SIP Express Router (SER) and a SER administrator.SERAdmin has an intuitive look and feel. SERAdmin provides control over many SER tasks such as: Start, Stop, Pause, Re-start, Monitor SER, Add User, Change Password and EmailId Delete User, Add Alias , Edit Alias etc. With this we can also use FIFO commands,Access controls,User Location. The objective of the SERAdminv03e is to support the ser-0.8.11 version also. This latest SER Administration Application handles both ser-0.8.10 and ser-0.8.11. What is new?. The following commands available in this release ------------------------------------------------ Ping(uri) - Pinging a URI Cisco_restart(uri) - Restart a Cisco Phone arg - Arguments of SER pwd - Present Working Directory t_uac_dlg - Initiate a Transaction you can download the components from http://developer.berlios.de/projects/seradmin For any feedback & help pls. contact xten_india(a)yahoo.com Regards, Team, Xten_india --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software

2 1

[Serusers] Outbound proxy
by Peter Hudec 31 Oct '03

31 Oct '03

hello, this is maybe a bit silly question, but. Outbound proxy is needed, when UA is behind FW. What is the function of this proxy (how can this help the UA to make calls) and what kind of software can do this. Can SER be also outbound proxy, or I have to use something else ? hudecof -- mail: [phudec(a)postel.sk] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg]

2 1

[Serusers] Gentoo
by danny adams 31 Oct '03

31 Oct '03

Is this latest version of Ser already intergrated on release of latest version......in Gentoo. Danny --------------------------------- Post your free ad now! Yahoo! Canada Personals

2 1