sr-users November 2004

sr-users@lists.kamailio.org

203 participants
386 discussions

[Serusers] MySQL Compatibility Questions

by Richard Machida

I had installed SER v 0.8.14 with MySQL v3 and it ran great. During the install of SerWeb, it mentioned in the documentation that MySQL v4 was required for some functions to work. After installing MySQL v4.1.7, SER won't run. It will run if all references to MySQL are removed from the ser.cfg. I tried: Drop the ser database and re-installed. recompiled SER Tried # ser -ddddddd -E -c -f /etc/ser/ser and got back "config file ok, exiting..." Is there a compatibility issue? Or something I'm missing. Thank you -- Richard Machida

20 years

[Serusers] Voicemail: Strange behavior if caller-user cancels the call

by Rafael J. Risco G.V.

Hi I need to solve this problem I´ve reported several times: please take a look into this report: http://lists.iptel.org/pipermail/serusers/2004-August/010930.html ...If the called-user belongs to the voicemail group and the caller-user cancels the call I see irregular behavior in both SER instances and No "Sip-Response-Code=487" is recorded in radius acc (just an start record without stop)... same problem if I use Asterisk as a voicemail system, please send me some advice. thank you Rafael -- rrgv

20 years

[Serusers] authentication problem

by Martin Bangiev

Hi all, I have ttrouble making IP phone (BudgeTone-100) + Ser + Asterisk work together with authentication I installed and setup the ser according to the instructions in the howto-s with enabled mysql authentication. The ser and the asterisk are working together on a single maschine (10.3.3.7). Ser is listening on port 5060 and asterisk is on port 5061. What I want is when the IP phone (10.10.0.13) get authenticated to be forwarded from ser to asterisk. Here is my route script (it's quite simple i think): route{ if (!proxy_authorize("asterisk", "subscriber")) { proxy_challenge("asterisk", "0"); break; }; forward(10.3.3.7, 5061); } here is a piece of the ser's output: 0(16841) check_nonce(): comparing [4190d1a0be145d87dc84c516175f8b46bc4923d6] and [4190d1a0be145d87dc84c516175f8b46bc4923d6] 0(16841) query="select password from subscriber where username='bangieff' AND domain='asterisk'" 0(16841) HA1 string calculated: fe6a6d50bebb95056a4a4cd70c12bf8d 0(16841) check_response(): Our result = '6362285d3385465321ce523620522056' 0(16841) check_response(): Authorization is OK 0(16841) save_rpid(): rpid value is '' 0(16841) check_via_address(10.10.0.13, 10.10.0.13, 0) 0(16841) Sending: when I remove the authorization part everything works just fine. Please somebody tell me where the problem could be Thanks in advance Martin Bangiev

20 years

[Serusers] Do I have to use jabber module?

by C Y

HI everyone I want to realize two users who log in the same SIP server with Windows Messenge to communicate each other.Now they can communicate through voice.But they cannot send messenges to others.Do i have to use jabber module to implement the function or there is an other way to realize it? thanks a lot. _________________________________________________________________ 享用世界上最大的电子邮件系统― MSN Hotmail。 http://www.hotmail.com

20 years

Re: [Serusers] error restarting ser

by Vinayak Krishan

Marian - Thanks much... sorted out the error.. In my enthusiasm, I forgot to load up the MySQL dynamic client lib.. It got past that error... Now for the next error.. I have MySQL 4.1.7 server, client, dynamic client lib, shared lib... When I try to start SER, the error I get is that the MySQL client authentication protocol does not match the MySQL server.. A Google search came back with something like upgrade the client versoin to better than 4.1.1.. Since I already have 4.1.7, that does not seem to be the solution... yet, just to be on the safe side, I changed the MySQL's /etc/my.cnf file to startup with the --old-password command.. No luck yet on getting SER to start Any suggesitons from anyone if they have seen this and how they got it working? Yes.. I have gone into the MySQL support site and seen the options there... Vinayak marian.dumitru@voic e-sistem.ro To: "Vinayak Krishan" <Vinayak.Krishan(a)relianceinfo.com> Sent by: cc: serusers(a)lists.iptel.org serusers-bounces@ip Subject: Re: [Serusers] error restarting ser tel.org Importance: Normal Sender's OU: Reliance |------------------| | [ ] Confidential | |------------------| 11/08/04 08:00 PM > Hi, > you can check your config file with "ser -ddddddd -E -c -f config_file" and you will get in the debug messages the line/position where the error is located. You should consider v0.8.14 - is a newer release. > Best regards, > Marian Dumitru > I'm installing SER 8.12 on a Fedora Box with MySQL 4.1.7-0... I've > followed > the steps in SER-howto completely until I get to the section down the > middle where it asks to modify the ser.cfg file.. After modifying it for > mysql, digest authentication and changing the uri to uri=~"vinayakk.local" > , the ser restart command comes back with a "error in config file" message > .. > > Could you let me know how I can find out what is causing this error? > > thanks > > _______________________________________________ > 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

20 years

[Serusers] Problem with specific phones only able to register to ipnumber and not domain..

by Claus Futtrup

Hi Guys, I have this problem with a few phones, that they can only use IP number for proxy adress and domain. So when they register to the proxy, the location database will have user@ipnumber and not user@domain (use_domain, 0). The problem is then as follows: that when a caller that registeres with the domain name are unable to call users that register with the domains IP number and vice versa. lookup("location") simply replies not found. Now I know why it happens, but is there a solution for this problem. Like only look at user and not user@domain or user@ipnumber? Kind Regards Claus Futtrup

20 years

[Serusers] SIP installation

by Domagoj Pehar

Hi everybody, I'm quite new in the SIP world and also Linux operating system. I read some instructions how to install SIP on Linux based PC (from some documents on web) but it didn't work. Does anybody have some document which describes step by step intallation of SIP server on the Linux OS (e.g. Rad Hat 3.0 ES or Rad Hat 9)? What are prerequisites for installing it? Does SER provide user grafical interface or I have to install additional application? Thanks Domagoj

20 years

[Serusers] Mediaproxy problem

by Ricardo Martinez

Hello List. Thanks for the answers about my last question. Now i have another problem. I installed mediaproxy to work with some NAT'd clients. When a NAT'd client calls to another client i only have voice in one way. This, at first sight, seems to be a configuration problem, but i'm using the same configuration that in my lab test and worked ok. So i'm a little bit confused about this situation. For the other hand the only thing added in this new configuration is a "exec" command that changes the R-URI for all the calls (add a prefix). I'm attaching part of the debug of the call and part of the debug of the mediaproxy (tail -f /var/log/messages). # U xx.xx.148.186:64443 -> xx.xx.154.35:5060 INVITE sip:2408196@sipproxy.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bKc0420a0ba45 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com> Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 5 INVITE Supported: timer, replaces Min-SE: 1800 Date: Tue, 14 Jun 2005 02:15:28 GMT User-Agent: AddPac SIP Gateway Contact: <sip:5555848110@192.168.0.191> Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 241 Max-Forwards: 70 v=0 o=5555848110 1118715328 1118715328 IN IP4 192.168.0.191 s=AddPac Gateway SDP c=IN IP4 192.168.0.191 t=1118715328 0 m=audio 23006 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 # U xx.xx.154.35:5060 -> xx.xx.148.186:64443 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bKc0420a0ba45;rport=64443;received=xx.xx.148. 186 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=6eb14299de2cc731d173c135064daec5.695 6 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 5 INVITE Proxy-Authenticate:Digest realm="sipproxy.mydomain.com", nonce="418faa143a3b169ae45516c762eb6effa17fa462", qop="auth" Server: Sip EXpress router(0.8.14-3 (i386/linux)) Content-Length: 0 Warning: 392 xx.xx.154.35:5060 "Noisy feedback tells: pid=12040 req_src_ip=xx.xx.148.186 req_src_port=64443 in_uri=sip:2408196@sipproxy.mydomain.com out_uri=sip:7072408196@sipproxy.mydomain.com via_cnt==1" . . # U xx.xx.148.186:64443 -> xx.xx.154.35:5060 ACK sip:2408196@sipproxy.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bKc0420a0ba45 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=6eb14299de2cc731d173c135064daec5.695 6 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 5 ACK Content-Length: 0 Max-Forwards: 70 # U xx.xx.148.186:64443 -> xx.xx.154.35:5060 INVITE sip:2408196@sipproxy.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bKc0420a0ba46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com> Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 INVITE Supported: timer, replaces Min-SE: 1800 Date: Tue, 14 Jun 2005 02:15:28 GMT User-Agent: AddPac SIP Gateway Contact: <sip:5555848110@192.168.0.191> Accept: application/sdp Proxy-Authorization: Digest username="1-12-2MPK8X2PK2", realm="sipproxy.mydomain.com", nonce="418faa143a3b169ae45516c762eb6effa17fa462", uri="sip:2408196@sipproxy.mydomain.com", qop=auth, nc=00000001, cnonce="c651d599", response="9f78e67834ea62cf90682f60b4aa7191", algorithm=MD5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 241 Max-Forwards: 70 v=0 o=5555848110 1118715328 1118715328 IN IP4 192.168.0.191 s=AddPac Gateway SDP c=IN IP4 192.168.0.191 t=1118715328 0 m=audio 23006 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 # U xx.xx.154.35:5060 -> xx.xx.148.186:64443 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bKc0420a0ba46;rport=64443;received=xx.xx.148. 186 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com> Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 INVITE Server: Sip EXpress router (0.8.14-3 (i386/linux)) Content-Length: 0 Warning: 392 xx.xx.154.35:5060 "Noisy feedback tells: pid=12037 req_src_ip=xx.xx.148.186 req_src_port=64443in_uri=sip:2408196@sipproxy.mydomain.com out_uri=sip:7072408196@xx.xx.148.231:5060 via_cnt==1" # U xx.xx.154.35:5060 -> xx.xx.148.231:5060 INVITE sip:7072408196@xx.xx.148.231:5060 SIP/2.0 Record-Route: <sip:2408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> Via: SIP/2.0/UDP xx.xx.154.35;branch=z9hG4bKcc3.a0569a31.0 Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com> Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 INVITE Supported: timer, replaces Min-SE: 1800 Date: Tue, 14 Jun 2005 02:15:28 GMT User-Agent: AddPac SIP Gateway Contact: <sip:5555848110@xx.xx.148.186:64443> Accept: application/sdp Proxy-Authorization: Digest username="1-12-2MPK8X2PK2", realm="sipproxy.mydomain.com", nonce="418faa143a3b169ae45516c762eb6effa17fa462", uri="sip:2408196@sipproxy.mydomain.com", qop=auth, nc=00000001, cnonce="c651d599", response="9f78e67834ea62cf90682f60b4aa7191", algorithm=MD5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 240 Max-Forwards: 69 P-hint: fixed NAT contact for request P-hint: H323-Side v=0 o=5555848110 1118715328 1118715328 IN IP4 192.168.0.191 s=AddPac Gateway SDP c=IN IP4 xx.xx.154.35 t=1118715328 0 m=audio 35024 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 # U xx.xx.148.231:5060 -> xx.xx.154.35:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP xx.xx.154.35;branch=z9hG4bKcc3.a0569a31.0 Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com> Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 Nov 8 12:12:08 sipvoiss mediaproxy[10717]: command request c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 192.168.0.191:23006:audio xx.xx.148.186 sipproxy.mydomain.com remote sipproxy.mydomain.com remote AddPac=20SIP=20Gateway info=from:5555848110@sipproxy.mydomain.com,to:2408196@sipproxy.mydomain.com, fromtag:c0420a0ba4,totag: Nov 8 12:12:08 sipvoiss mediaproxy[10717]: session c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191: started. listening on xx.xx.154.35:35024 Nov 8 12:12:08 sipvoiss mediaproxy[10717]: command execution time: 1.68 ms # U xx.xx.148.231:5060 -> xx.xx.154.35:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP xx.xx.154.35;branch=z9hG4bKcc3.a0569a31.0 Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 INVITE User-Agent: AddPac SIP Gateway Contact: sip:7072408196@xx.xx.148.231 Content-Length: 0 Record-Route: <sip:2408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> # U xx.xx.154.35:5060 -> xx.xx.148.186:64443 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 INVITE User-Agent: AddPac SIP Gateway Contact: sip:7072408196@xx.xx.148.231 Content-Length: 0 Record-Route: <sip:2408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> Nov 8 12:12:10 sipvoiss mediaproxy[10717]: session c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Nov 8 12:12:10 sipvoiss mediaproxy[10717]: session c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191: ended (did timeout). # U xx.xx.148.231:5060 -> xx.xx.154.35:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.154.35;branch=z9hG4bKcc3.a0569a31.0 Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 INVITE User-Agent: AddPac SIP Gateway Contact: sip:7072408196@xx.xx.148.231 Content-Type: application/sdp Content-Length: 208 Record-Route: <sip:2408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> v=0 o=7072408196 1099959234 1099959234 IN IP4 xx.xx.148.231 s=AddPac Gateway SDP c=IN IP4 xx.xx.148.231 t=1099959234 0 m=audio 23300 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 # U xx.xx.154.35:5060 -> xx.xx.148.186:64443 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 INVITE User-Agent: AddPac SIP Gateway Contact: sip:7072408196@xx.xx.148.231 Content-Type: application/sdp Content-Length: 208 Record-Route: <sip:2408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> v=0 o=7072408196 1099959234 1099959234 IN IP4 xx.xx.148.231 s=AddPac Gateway SDP c=IN IP4 xx.xx.148.231 t=1099959234 0 m=audio 23300 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 # U xx.xx.148.186:64443 -> xx.xx.154.35:5060 ACK sip:7072408196@xx.xx.148.231 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bKc0420a0ba46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 ACK Route: <sip:2408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> Content-Length: 0 Max-Forwards: 70 # U xx.xx.154.35:5060 -> xx.xx.148.231:5060 ACK sip:7072408196@xx.xx.148.231 SIP/2.0 Record-Route: <sip:7072408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> Via: SIP/2.0/UDP xx.xx.154.35;branch=0 Via: SIP/2.0/UDP 192.168.0.191:5060;received=xx.xx.148.186;branch=z9hG4bKc0420a0ba46 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 6 ACK Content-Length: 0 Max-Forwards: 69 P-hint: USRLOC Nov 8 12:12:19 sipvoiss mediaproxy[10717]: command lookup c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 xx.xx.148.231:23300:audio xx.xx.148.231 sipproxy.mydomain.com remote sipproxy.mydomain.com unknown AddPac=20SIP=20Gateway info=from:5555848110@sipproxy.mydomain.com,to:2408196@sipproxy.mydomain.com, fromtag:c0420a0ba4,totag:b741e2b9a4 Nov 8 12:12:19 sipvoiss mediaproxy[10717]: warning: trying to lookup session with non-existent id: 'c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191' Nov 8 12:12:19 sipvoiss mediaproxy[10717]: command execution time: 0.25 ms # U xx.xx.148.186:64443 -> xx.xx.154.35:5060 BYE sip:7072408196@xx.xx.148.231 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.191:5060;branch=z9hG4bKc0420a0ba47 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 7 BYE Route: <sip:2408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> Date: Tue, 14 Jun 2005 02:15:49 GMT User-Agent: AddPac SIP Gateway Contact: <sip:5555848110@192.168.0.191> Content-Length: 0 Max-Forwards: 70 # U xx.xx.154.35:5060 -> xx.xx.148.231:5060 BYE sip:7072408196@xx.xx.148.231 SIP/2.0 Record-Route: <sip:7072408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> Via: SIP/2.0/UDP xx.xx.154.35;branch=z9hG4bKdc3.ee3b7823.0 Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a47 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 7 BYE Date: Tue, 14 Jun 2005 02:15:49 GMT User-Agent: AddPac SIP Gateway Contact: <sip:5555848110@xx.xx.148.186:64443> Content-Length: 0 Max-Forwards: 69 P-hint: fixed NAT contact for request P-hint: USRLOC # U xx.xx.148.231:5060 -> xx.xx.154.35:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.154.35;branch=z9hG4bKdc3.ee3b7823.0 Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a47 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 7 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 Record-Route: <sip:7072408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> # U xx.xx.154.35:5060 -> xx.xx.148.186:64443 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.191:5060;rport=64443;received=xx.xx.148.186;branch=z9hG4bKc0420a0b a47 From: <sip:5555848110@sipproxy.mydomain.com>;tag=c0420a0ba4 To: <sip:2408196@sipproxy.mydomain.com>;tag=b741e2b9a4 Call-ID: c03dae42-d936-0aa2-800b-0002a400f1e9(a)192.168.0.191 CSeq: 7 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 Record-Route: <sip:7072408196@xx.xx.154.35;ftag=c0420a0ba4;lr=on> As you can see the mediaproxy session is ended by timeout. I don't understand why. Can someone give me a hand here? Thanks in advance Ricardo Martinez Ogalde.-

20 years

[Serusers] Ser + Asterisk

by Walter Willis

the asterisk and ser can run in one machine? --------------------------------- Do You Yahoo!? Todo lo que quieres saber de Estados Unidos, América Latina y el resto del Mundo. Visíta Yahoo! Noticias.

20 years

Re: [Serusers] ser+asterisk again

by Marian Dumitru

Hi Graziano, most probably your routing is different fro INVITEs and CANCELs - check if you don't have in your script some specific only for INVITEs and not for CANCEL. CANCEL must be routed all the time on the same criteria as INVITE. Best regards, Marian Graziano Poretti wrote: > hi marianu > it works .. thnx. the only problem now is that if i end the conversation > from my ip phone before that on the other side will reply, on the other > side doesn't stop ringing ... is this correct or it's a fault (may be i > have to change something on my ser/asterisk) > > sorry .. i'm a newbie here ... may be that this was a silly question, > but thnx in any case. > > thnx > > > > RAINBOW TEAM Graziano Poretti Rainbow Product Manager Due Metri s.a.s. > graziano(a)duemetri.com tel: ++39 0184 42163 Signature powered by Plaxo > Want a signature like this? Add me to your address book... > ----- Original Message ----- From: "Marian Dumitru" > <marian.dumitru(a)voice-sistem.ro> > To: "Graziano Poretti" <graziano(a)duemetri.com> > Cc: <serusers(a)lists.iptel.org> > Sent: Saturday, November 06, 2004 11:31 AM > Subject: Re: [Serusers] ser+asterisk again > > >> Hi Graziano >> >> if you want to divert the calls with "00" from ser to a asterisk, you >> just need to add the following lines in your ser.cfg (probably >> somewhere inside "uri==myself" if consition, just before >> lookup("location") - if you are using standard script) : >> >> if (uri=~"sip:00") { >> t_relay_to_udp("asterisk_IP","asterisk_port"); >> break; >> } >> >> Best regards, >> Marian Dumitru >> >> >> Graziano Poretti wrote: >> >>> hi >>> we have 1 ser + 1 asterisk on different computers >>> we want to forward incoming calls from the server with fixed prefix >>> (eg 00) to asterisk. >>> i tried different solutions but how is the connection logic between >>> ser and asterisk? should i have a registered user in asterisk which >>> collects all the traffic coming from ser or not? how can i setup the >>> ser and asterisk to have then "talking"? >>> thnx in advance for help >>> Graziano >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Voice Sistem >> http://www.voice-sistem.ro >> > -- Voice Sistem http://www.voice-sistem.ro

20 years

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sr-users November 2004