sr-users March 2004

sr-users@lists.kamailio.org

145 participants
286 discussions

[Serusers] ser logs
by Muhammad Nasim 12 Mar '04

12 Mar '04

I understand that by default ser doesn't write much log output and what is does ends up in /var/log/messages via the syslog facility. I also understand that one can get detailed logs printed to stderr by changing some settings in ser.cfg and restarting ser. Does anyone have a suggestion as to the best way to get detailed logs dumped to a file so that I can tail them etc?

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Re: [Serusers] SIP call to ISDN subscriber
by Jiri Kuthan 12 Mar '04

12 Mar '04

most likely caused by a misrouted ACK. -jiri At 10:36 AM 3/12/2004, Manuel Goertz wrote: >Hi all, > >I have a problem calling from a sipset to a ISDN subscriber over >a CISCO 1760 GW. >The following setup is used. >UA ---> GW ---> ISDN >The UA is a sipset, the GW is the CISCO 1760 with ISDN BRI interface >and a standard ISDN subscriber. >The UA is registered with the SER server. >All numeric userparts of the SIP URI are rewritten and routed to the GW. >The GW's BRI interface is connected to the PSTN. >The call signaling seems to work as the SIP phone indicates ringing >and the ISDN phone is ringing. After picking up the hook of the ISDN >phone the UA shows "In Call". But after a second the call is >terminated. The log shows that the GW sends to both side the call >termination messages. TX -> DISCONNECT "Normal Call Clearing" to the >ISDN side and a BYE message to the SIP side. >The signaling in short: > >UA GW ISDN >INVITE -> | > <- 100 Try > | TX -> SETUP > | RX <- CALL_PROC > | RX <- ALERTING > <- 183 Sess | > | RX <- CONNECT > | TX -> CONNECT_ACK > <- 200 OK | > Milliseconds later ! > | TX -> DISCONNECT > | RX <- RELEASE > | TX -> RELEASE_COMP > <- BYE | >200 OK -> | > > >Any hints how to solve this problem. > >Thanks > > Manuel > > > > > > >-- >+KOM----------------------------------------------------------------+ >|Manuel Görtz Merckstrasse 25| >|Darmstadt University of Technology 64283 Darmstadt, Germany| >|Multimedia Communications Tel: (+49) 6151 16-5175| >|Multimedia Networking & Distribution Fax: (+49) 6151 16-6152| >+----------------------------------------------------------------KOM+ > >_______________________________________________ >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] SER installation
by M. B. 12 Mar '04

12 Mar '04

I just started with SER. I downloaded and compiled SER with MYSQL auth. Everything worked fine. I then compiled and installed the radius server and made sure it works. Then I compiled SER with Radius options, and then changed the SER configuration file based on the howto file. But when I run sectl start I get the following error: #serctl start Starting SER : cat: /var/run/ser.pid: No such file or directory started pid() and ser is not running. If I comment out everything related to Radius, everything runs. I probably have a bad configuration. Can someone please post a copy of a working ser configuration file with Radius authentication, authorization, and accounting and a couple of outside gateway routing example all enabled? __________________________________ Do you Yahoo!? Yahoo! Search - Find what you�re looking for faster http://search.yahoo.com

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[Serusers] Outbound dialing setup
by Doug R 12 Mar '04

12 Mar '04

I'm using the script from http://lists.iptel.org/pipermail/serusers/2004-February/005996.html (the first ser.cfg). I'm trying to figure out where to include my SIP outgoing calls to my Asterisk PSTN server. For example, let's say I want to transfer a SIP call for a local XXX-XXXX number to pstn.xyz.com port 5050. How would I set that up? Additionally, I'm rather confused as to which section I would put a dialplan into for say connecting to FWD users, etc. Again, like before if I wanted to send a certain call to a certain termination point, let's say 407-xxx-xxxx, I want to send that to one IP, and 954-xxx-xxxx I want to sent to a different one. Where are those statements supposed to be? Thanks a lot...

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[Serusers] SIP call to ISDN subscriber
by Manuel Goertz 12 Mar '04

12 Mar '04

Hi all, I have a problem calling from a sipset to a ISDN subscriber over a CISCO 1760 GW. The following setup is used. UA ---> GW ---> ISDN The UA is a sipset, the GW is the CISCO 1760 with ISDN BRI interface and a standard ISDN subscriber. The UA is registered with the SER server. All numeric userparts of the SIP URI are rewritten and routed to the GW. The GW's BRI interface is connected to the PSTN. The call signaling seems to work as the SIP phone indicates ringing and the ISDN phone is ringing. After picking up the hook of the ISDN phone the UA shows "In Call". But after a second the call is terminated. The log shows that the GW sends to both side the call termination messages. TX -> DISCONNECT "Normal Call Clearing" to the ISDN side and a BYE message to the SIP side. The signaling in short: UA GW ISDN INVITE -> | <- 100 Try | TX -> SETUP | RX <- CALL_PROC | RX <- ALERTING <- 183 Sess | | RX <- CONNECT | TX -> CONNECT_ACK <- 200 OK | Milliseconds later ! | TX -> DISCONNECT | RX <- RELEASE | TX -> RELEASE_COMP <- BYE | 200 OK -> | Any hints how to solve this problem. Thanks Manuel -- +KOM----------------------------------------------------------------+ |Manuel Görtz Merckstrasse 25| |Darmstadt University of Technology 64283 Darmstadt, Germany| |Multimedia Communications Tel: (+49) 6151 16-5175| |Multimedia Networking & Distribution Fax: (+49) 6151 16-6152| +----------------------------------------------------------------KOM+

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[Serusers] Unable to register - Noisy feedback
by Alan Litster 12 Mar '04

12 Mar '04

Hi, We've got an intertex IX66 SIP User agent that we are trying to register on SER (0.8.12 CVS). I've attached the ngrep output of the registeration attempt, it reports "Noisy feedback tells". However it works fine with other clients. (We're using radius based authentication) I've tried adding xlog commands to ser.cfg, though nothing appears in the log, though ngrep picks it up... What could the problem be ?? Regards, Alan # U 82.32.34.228:5060 -> 1.2.3.81:5060 REGISTER sip:sip.com SIP/2.0..Via: SIP/2.0/UDP 82.32.34.228:5060;bran ch=z9hG4bK0fc1a0e27d3f673ab0e7b18b82c39281.0..Via: SIP/2.0/UDP 0.0.0.0;xdat a=Qv4u+bUNyCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sipcal l.co.uk..To: sip:12349000@sip.com..Call-ID: 000ab761-6341000e-3ea49 b31-20169200@192.168.1.105..Date: Thu, 11 Mar 2004 17:09:21 GMT..CSeq: 101 REGISTER..User-Agent: CSCO/4..Contact: <sip:kTHowCXnSy7o@82.32.34.228>;expi res=3600..Content-Length: 0..Expires: 3600.... # U 1.2.3.81:5060 -> 82.32.34.228:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 82.32.34.228:5060;branch=z9hG4bK 0fc1a0e27d3f673ab0e7b18b82c39281.0..Via: SIP/2.0/UDP 0.0.0.0;xdata=Qv4u+bUN yCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sip.com..T o: sip:12349000@sip.com;tag=cc34cfa6d75af81b0409220a2ba8e06a.0b47.. Call-ID: 000ab761-6341000e-3ea49b31-20169200@192.168.1.105..CSeq: 101 REGIS TER..WWW-Authenticate: Digest realm="sip.com", nonce="40509e6d0eeb5ff c4083b4ba14c7d03479746aeb", qop="auth"..Server: Sip EXpress router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 1.2.3.81:5060 "Noisy fe edback tells: pid=22432 req_src_ip=82.32.34.228 req_src_port=5060 in_uri=s ip:sip.com out_uri=sip:sip.com via_cnt==2".... # U 82.32.34.228:5060 -> 1.2.3.81:5060 REGISTER sip:sip.com SIP/2.0..Via: SIP/2.0/UDP 82.32.34.228:5060;bran ch=z9hG4bK60a2f914cc748de2b0e7b18b82c39281.0..Via: SIP/2.0/UDP 0.0.0.0;xdat a=Qv4u+bUNyCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sipcal l.co.uk..To: sip:12349000@sip.com..Call-ID: 000ab761-6341000e-3ea49 b31-20169200@192.168.1.105..Date: Thu, 11 Mar 2004 17:09:21 GMT..CSeq: 102 REGISTER..User-Agent: CSCO/4..Contact: <sip:kTHowCXnSy7o@82.32.34.228>;expi res=3600..Authorization: Digest username="12349000",realm="sip.com" ,uri="sip:sip.com",response="a597e49eb1578fad3ed27ec38beb460c",nonce= "40509e6d0eeb5ffc4083b4ba14c7d03479746aeb",cnonce="55a1ffcb",qop=auth,nc=00 000001,algorithm=md5..Content-Length: 0..Expires: 3600.... # U 82.32.34.228:5060 -> 1.2.3.81:5060 REGISTER sip:sip.com SIP/2.0..Via: SIP/2.0/UDP 82.32.34.228:5060;bran ch=z9hG4bK60a2f914cc748de2b0e7b18b82c39281.0..Via: SIP/2.0/UDP 0.0.0.0;xdat a=Qv4u+bUNyCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sipcal l.co.uk..To: sip:12349000@sip.com..Call-ID: 000ab761-6341000e-3ea49 b31-20169200@192.168.1.105..Date: Thu, 11 Mar 2004 17:09:21 GMT..CSeq: 102 REGISTER..User-Agent: CSCO/4..Contact: <sip:kTHowCXnSy7o@82.32.34.228>;expi res=3600..Authorization: Digest username="12349000",realm="sip.com" ,uri="sip:sip.com",response="a597e49eb1578fad3ed27ec38beb460c",nonce= "40509e6d0eeb5ffc4083b4ba14c7d03479746aeb",cnonce="55a1ffcb",qop=auth,nc=00 000001,algorithm=md5..Content-Length: 0..Expires: 3600.... # U 1.2.3.81:5060 -> 82.32.34.228:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 82.32.34.228:5060;branch=z9hG4bK 60a2f914cc748de2b0e7b18b82c39281.0..Via: SIP/2.0/UDP 0.0.0.0;xdata=Qv4u+bUN yCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sip.com..T o: sip:12349000@sip.com;tag=cc34cfa6d75af81b0409220a2ba8e06a.ecd2.. Call-ID: 000ab761-6341000e-3ea49b31-20169200@192.168.1.105..CSeq: 102 REGIS TER..WWW-Authenticate: Digest realm="sip.com", nonce="40509e6f4a04de1 ce336dd980cc5962ca10bb63b", qop="auth"..Server: Sip EXpress router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 1.2.3.81:5060 "Noisy fe edback tells: pid=22433 req_src_ip=82.32.34.228 req_src_port=5060 in_uri=s ip:sip.com out_uri=sip:sip.com via_cnt==2".... # U 1.2.3.81:5060 -> 82.32.34.228:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 82.32.34.228:5060;branch=z9hG4bK 60a2f914cc748de2b0e7b18b82c39281.0..Via: SIP/2.0/UDP 0.0.0.0;xdata=Qv4u+bUN yCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sip.com..T o: sip:12349000@sip.com;tag=cc34cfa6d75af81b0409220a2ba8e06a.ecd2.. Call-ID: 000ab761-6341000e-3ea49b31-20169200@192.168.1.105..CSeq: 102 REGIS TER..WWW-Authenticate: Digest realm="sip.com", nonce="40509e711121e42 910dfd8d6d4ee741bf6615d18", qop="auth"..Server: Sip EXpress router (0.8.12 (i386/linux))..Content-Length: 0..Warning: 392 1.2.3.81:5060 "Noisy fe edback tells: pid=22434 req_src_ip=82.32.34.228 req_src_port=5060 in_uri=s ip:sip.com out_uri=sip:sip.com via_cnt==2".... ------------------------------------------------------------------------------------------------------- This email, and any files transmitted with it, is copyright and may contain confidential information. The contents are intended for the use of the addressee(s) only. Unauthorized use may be unlawful. If you receive this email by mistake, please advise sender immediately. The views of the author may not necessarily constitute the views of Telco Electronics Limited. Nothing in this mail shall bind Telco Electronics Limited in any contract or obligation. 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[Serusers] 407 proxy authentication message after BYE
by Samy Touati 11 Mar '04

11 Mar '04

Hi, I'm using a WLAN sip phone, and I'm connecting to my SIP server from different locations. In some locations, when I connect, place a call, then hang up (from the sip phone) ser receives my BYE message but send me a 407 proxy authentication required message, this message is not answered by my sip phone. >From other locations, the BYE message is received by ser and relayed to he other end point, without problems. In which case a proxy authentication is required after a BYE ? Thanks. Samy.

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[Serusers] SIP VoIP Problem & Shorewall
by Andrei Verovski 11 Mar '04

11 Mar '04

Hi, I am struggling to get SIP (session initiation protocol) VoIP software (SIP Express Router + XTen/KPhone/Linphone) and hardware (Grandstream SIP IP Phone) to work with Shorewall. The best result I have achieved is when I can call (or someone call me), hear sound, but my talk is not transmitted for weird reason. I have quite standard configuration - two-card Linux PC with shorewall (one static external IP + static local IPs 192.168.0.xxx), opened ports are 5060 tcp/udp, 5004 udp (for Grandstream), and 8000-8020 udp (for XTen, Linphone). SER itself is hosted by the independent provider, and other people who have another router/firewall software do not have similar problem. It looks like something is wrong with NAT or masquarading. May be someone will be kind to e-mail (or post on this mailing list) their Shorewall configs (masq, nat, rules, etc.)? My e-mail is andreil1(a)starlett.lv Thanks in advance. ********************************************* * Best Regards --- Andrei Verovski * * Personal Home Page * http://snow.prohosting.com/guru4mac/ * Mac, Linux, DTP, Development, IT WEB Site *********************************************

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[Serusers] Post installation Problems with SER and MYSQL.
by Shahzad Sadiq 11 Mar '04

11 Mar '04

Hi, I ve recently started with SER. So far I ve managed to install MYSQL and SER on REDHAT LINUX ver 9. I can also start the MYSQL Server and SER. The problem I am facing now is that when ever i go to /usr/sbin and give the command ./ser_mysql.sh <variable> to create the mysql table it gives me error messageas follows: this is the message with ./ser_mysql.sh create>> > creating database ser ... > ./ser_mysql.sh: line 80: mysql: command not found. With"reinstall" gives me different error message which is as follows: > ./ser_mysql.sh: line 102: mysqldump: command not found > ser backup dump failed. Can anyone help me on this I know i am missing something but I am not quite sure.And can any one please tell me how to configure Messenger service as well. Any help on this I'll be really greatful. Thanks shahzad

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[Serusers] ser/rtp exerciser
by Alex Bligh 11 Mar '04

11 Mar '04

Before I reinvent the wheel and go write it, is there a utility which will simulate making & receiving calls (i.e. SIP conversations, **plus RTP**) in scale (to answer a question like "will my SIP+RTPproxy deployment scale for {100,1000,10000,100000} users using the following parameters for average call duration, % phones in use etc." If not, anyone have any recommendations for a (free) stack that does a decent UAC/UAS job, including audio, and will run without GUI? Alex

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sr-users March 2004