sr-users March 2006

sr-users@lists.kamailio.org

223 participants
506 discussions

Re: [Serusers] rtpproxy
by Andrey Kouprianov 02 Mar '06

02 Mar '06

Im not sure about it, cause I never had chance to play with PSTN numbers properly. There's something about DID's on page 58 of the document. On 3/3/06, roger leszczynski <rogerles(a)gmail.com> wrote: > i found it, thanks, now the question is, say i have 100 concurrent calls, > and i want to update the configuration with a new DID, how can i avoid > restarting SER to make these changes take places....has anyone made a > similar module to reload in asterisk? Or how can i go about using a > database to implement routing instead? > > > On 3/2/06, Andrey Kouprianov <andrey.kouprianov(a)gmail.com > wrote: > > > would "getting started" document from www.onsip.org help? they have a > good explanation on how to configure SER with RTPProxy... > > On 3/2/06, roger leszczynski <rogerles(a)gmail.com> wrote: > > Anyone use the rtp proxy module yet? I am looking to proxy rtp of my > > carrier who sends out calls to me, so if anyone has example configurations > > that work i would appreciate it! > > > > _______________________________________________ > > 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 > > >

1 0

[Serusers] rtpproxy
by roger leszczynski 02 Mar '06

02 Mar '06

Anyone use the rtp proxy module yet? I am looking to proxy rtp of my carrier who sends out calls to me, so if anyone has example configurations that work i would appreciate it!

2 1

[Serusers] Redirect a sip outbound requests to a sip proxy
by hgaillac-sip＠yahoo.fr 02 Mar '06

02 Mar '06

Hi all, Is there a solution to solve this ? ASTERISK 1.2.4 || Internet===SER/OPENSER=====Nat==[private net] || sip agents rtpproxy/mediaproxy Sip agents use SER/OPENSER as an outbound sip proxy and asterisk as a registar server, pbx functions, ... SER/OPENSER look for domains in URI. if domains are handled by SER/OPENSER so requests are forwarded to ASTERISK. If domain in URI is not handled by SER/OPENSER and/or ASTERISK I wish the request is sent to Asterisk before to send back it to SER/OPENSER. in other word i wish to define SER/OPENSER server as oubound proxy for Astersik. Is it Possible ? Harry ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

1 0

[Serusers] serctl problem
by frankie chiang 02 Mar '06

02 Mar '06

Dear all, Please let me know what happen. Just create a new database by ser_mysql.sh script. # serctl add 200 200 200(a)test.com MySql password: error: 400; check if you use aliases in SER Best Regard, __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] using SER as a frontend for multiple asterisk servers?
by Roy Sigurd Karlsbakk 02 Mar '06

02 Mar '06

hi does anyone have a source of documentation that can help me setting up the following? one SER box (with a potential additional box for failover) should - Accept REGISTER and so on and just balance these over n asterisk boxes in behind. No authentication should be done on the SER box. - Accept INVITE and distribute this linearly to n asterisk boxes behind, starting with box 1, allow up to n calls to each box (each box having n PRI/E1s). - Accept INVITE from the asterisk boxes and forward these to the client - All RTP traffic should be proxied locally All help appreciated :) roy

1 0

Re: [Serusers] classkit.so module php5
by Kim Culhan 02 Mar '06

02 Mar '06

On Wed, March 1, 2006 5:01 pm, Ofer Zaarur said: > I have problem getting classkit module to set serweb correctly using > php5. it looks like this issue had been raise before. please let me know > where I can get it. You may find some problems running serweb with PHP5, much easier to run PHP4 as it has classkit. If you really want to try here is an example for building classkit at /usr/local/src: Get the source for php5 and untar to get: /usr/local/src/php5/php-5.1.2 Get the source for classkit from: http://pecl.php.net/package/classkit cd /usr/local/src/php5/php-5.1.2/ext and untar the source from the pecl web site for classkit in the directory above and rename the directory from classkit-0.4 to classkit. Then cd .. to the top level of the php-5.1.2 build system and run: phpize && ./configure --with-classkit cd to ext/classkit and you will now find a complete build tree run configure and make and you will get classkit.dsp, classkit.la, classkit.lo When you type make install it will install classkit.so in /usr/local/lib/php/20050922 The shared library version with the .so suffix is created in the above step. Hope this helps -kim -- w8hdkim(a)gmail.com

2 1

[Serusers] permission module
by Hakan YASTI 02 Mar '06

02 Mar '06

Hi, I am trying to use permissions module. At configuration side everything seems ok. when an invite comes to ser, gets the source ip of the invite and select a query from the trusted table. But every time, when correlates the from_pattern and proto, they mismatch ( I SUPPOSE ). Is there any trick at from_pattern ? I have tried so many values like .* , sip:.*,empty,^sip:.* and etc. Here is my ser.cfg, logs. Hakan. loadmodule "/usr/local/lib/ser/modules/permissions.so" # -- permissions parameters -- # modparam("permissions", "db_url", "postgres://xxxxx:xxxx@ip_address/xxxxx") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "source_col", "src_ip") modparam("permissions", "proto_col", "proto") modparam("permissions", "from_col", "from_pattern") if (method=="INVITE") { if (!allow_trusted()) { log("THE IP ADDRESS IS NOT ALLOWED"); sl_send_reply("403","ONLY REGISTERED USERS ALLOWED "); break; } ########### SER LOG ############# 2(69289) SIP Request: 2(69289) method: 2(69289) uri: 2(69289) version: 2(69289) parse_headers: flags=1 2(69289) Found param type 232, = ; state=16 2(69289) end of header reached, state=5 2(69289) parse_headers: Via found, flags=1 2(69289) parse_headers: this is the first via 2(69289) After parse_msg... 2(69289) preparing to run routing scripts... 2(69289) parse_headers: flags=128 2(69289) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(69289) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 2(69289) parse_headers: flags=8 2(69289) DEBUG: add_param: tag=1c681629589 2(69289) end of header reached, state=29 2(69289) grep_sock_info - checking if host==us: 9==10 && [gi.com.tr] == [84.51.32.8] 2(69289) grep_sock_info - checking if port 5060 matches port 5060 2(69289) grep_sock_info - checking if host==us: 9==9 && [gi.com.tr] == [127.0.0.1] 2(69289) grep_sock_info - checking if port 5060 matches port 5060 2(69289) grep_sock_info - checking if host==us: 9==10 && [gi.com.tr] == [84.51.32.8] 2(69289) grep_sock_info - checking if port 5060 matches port 5060 2(69289) grep_sock_info - checking if host==us: 9==9 && [gi.com.tr] == [127.0.0.1] 2(69289) grep_sock_info - checking if port 5060 matches port 5060 2(69289) val2str(): converting 84.51.32.26, 11 2(69289) PG[217] str2valp got string udp 2(69289) PG[217] str2valp got string sip:90224765321@gi.com.tr 2(69289) NOT TRUSTED IP 2(69289) parse_headers: flags=4 2(69289) end of header reached, state=9 2(69289) DEBUG: get_hdr_field: [40]; uri=[sip:02124440111@gi.com.tr;user=phone] 2(69289) DEBUG: to body [ ] 2(69289) parse_headers: flags=-1 2(69289) get_hdr_field: cseq : <1> 2(69289) DEBUG: get_hdr_body : content_length=267 2(69289) found end of header 2(69289) check_via_address(84.51.32.26, 84.51.32.26, 1) 2(69289) DEBUG:destroy_avp_list: destroying list 0x0 2(69289) receive_msg: cleaning up 3(69290) SIP Request: 3(69290) method: 3(69290) uri: 3(69290) version: 3(69290) parse_headers: flags=1 3(69290) Found param type 232, = ; state=16 3(69290) end of header reached, state=5 3(69290) parse_headers: Via found, flags=1 3(69290) parse_headers: this is the first via 3(69290) After parse_msg... 3(69290) preparing to run routing scripts... 3(69290) parse_headers: flags=4 3(69290) DEBUG: add_param: tag=26a82380ee921ee699cdfa26683b3165.bcf6 3(69290) end of header reached, state=29 3(69290) DEBUG: get_hdr_field: [82]; uri=[sip:02124440111@gi.com.tr;user=phone] 3(69290) DEBUG: to body [] 3(69290) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 3(69290) DEBUG:destroy_avp_list: destroying list 0x0 3(69290) receive_msg: cleaning up ######### NGREP LOGS ################ U 84.51.32.26:5060 -> 84.51.32.8:5060 INVITE sip:02124440111@gi.com.tr;user=phone SIP/2.0. Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. Max-Forwards: 70. From: "902247654321" ;tag=1c681629589. To: . Call-ID: 395527579dNhC(a)84.51.32.26. CSeq: 1 INVITE. Contact: . Supported: em,100rel,timer,replaces,path. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Session-Expires: 3600. Min-SE: 90. User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371. Content-Type: application/sdp. Content-Length: 267. . v=0. o=AudiocodesGW 440639 653740 IN IP4 84.51.32.26. s=Phone-Call. c=IN IP4 84.51.32.26. t=0 0. m=audio 4010 RTP/AVP 18 8 96. a=rtpmap:18 g729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 pcma/8000. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-15. a=ptime:40. a=sendrecv. # U 84.51.32.8:5060 -> 84.51.32.26:5060 SIP/2.0 403 ONLY REGISTERED USERS..... Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. From: "902247654321" ;tag=1c681629589. To: ;tag=26a82380ee921ee699cdfa26683b3165.bcf6. Call-ID: 395527579dNhC(a)84.51.32.26. CSeq: 1 INVITE. Server: Sip EXpress router (0.9.3 (i386/freebsd)). Content-Length: 0. Warning: 392 84.51.32.8:5060 "Noisy feedback tells: pid=69289 req_src_ip=84.51.32.26 req_src_port=5060 in_uri=sip:02124440111@gi.com.tr;user=phone out_uri=sip:02124440111@gi.com.tr;user=phone via_cnt==1". . # U 84.51.32.26:5060 -> 84.51.32.8:5060 ACK sip:02124440111@gi.com.tr;user=phone SIP/2.0. Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. Max-Forwards: 70. From: "902247654321" ;tag=1c681629589. To: ;tag=26a82380ee921ee699cdfa26683b3165.bcf6. Call-ID: 395527579dNhC(a)84.51.32.26. CSeq: 1 ACK. Contact: . Supported: em,timer,replaces,path. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371. Content-Length: 0. ################################# Tried another value for from_pattern... 3(69290) val2str(): converting 84.51.32.26, 11 3(69290) PG[217] str2valp got string udp 3(69290) PG[217] str2valp got string "90224765321" select * from trusted; src_ip | proto | from_pattern -------------+-------+--------------- 84.51.32.25 | udp | "90224765321" 84.51.32.26 | udp | "90224765321" (2 rows) .

2 3

[Users] t_relay and acc logging
by Script Head 02 Mar '06

02 Mar '06

Hello, I am struggling to make sense of acc logging when combined with forwarding a call. My openser.cfg is below this in message. Every time I place a call from a softphone that registeres to SER, I get 30+ records in acc, most of them INVITEs and some are ACKs and there are no BYEs. I can't seem to find any documentation that would explain to me what I am doing wrong. Someone, please put me on the right path. route{ setflag(3); if(method=="REGISTER") { if (!www_authorize("domain.com", "subscriber")) { www_challenge("richmedium.com", "0"); return; }; save("location"); return; } if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); return; }; if(method=="INVITE") { if (!www_authorize("richmedium.com", "subscriber")) { www_challenge("richmedium.com", "0"); return; }; } if (uri=~"^sip:1[0-9]*{10}@192.168.1.10") { forward(192.168.1.10,5061); } else { sl_send_reply("404", "Use prefix 1 to dial"); } t_relay(); }

2 1

[Users] UDP -> TCP BYE goes astray
by Dr. Rodney G. McDuff 02 Mar '06

02 Mar '06

I have an * server at XXX.XXX.2.46 and an opener 1.0.1 server at XXX.XXX.2.40 and a Windows messenger 5.1 UA at YYY.YYY.152.57. When WM5 is configured to use UDP a BYE transmitted my the * server is properly transmitted to the WM5 UA via the Openser proxy. All is good. > U 2006/03/02 15:17:27.332828 XXX.XXX.2.46:5060 -> XXX.XXX.2.40:5060 > BYE sip:YYY.YYY.152.57:16478 SIP/2.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK6a5f9584;rport. > Route: <sip:XXX.XXX.2.40;ftag=8400e0fa7cbe4cb5a23e935b7d64c1f3;lr=on>. > From: <sip:55555@domain.net>;tag=as25ec0bbf. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=8400e0fa7cbe4cb5a23e935b7d64c1f3;epid=1339bc4029. > Contact: <sip:55555@XXX.XXX.2.46>. > Call-ID: 45ddd129b23b48e791d5f82093a130a1. > CSeq: 102 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > > > U 2006/03/02 15:17:27.344137 XXX.XXX.2.40:5060 -> YYY.YYY.152.57:16478 > BYE sip:YYY.YYY.152.57:16478 SIP/2.0. > Record-Route: <sip:XXX.XXX.2.40;ftag=as25ec0bbf;lr=on>. > Via: SIP/2.0/UDP XXX.XXX.2.40;branch=z9hG4bK9ffb.cfbffdb2.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK6a5f9584;rport=5060. > From: <sip:55555@domain.net>;tag=as25ec0bbf. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=8400e0fa7cbe4cb5a23e935b7d64c1f3;epid=1339bc4029. > Contact: <sip:55555@XXX.XXX.2.46>. > Call-ID: 45ddd129b23b48e791d5f82093a130a1. > CSeq: 102 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 69. > Content-Length: 0. > However when the WM5 UA is configured to use TCP the RURI over written by the first Route header and the BYE goes nowhere with a 404 error. Why is this so and how can it be overcome. > > U 2006/03/02 15:07:59.622462 XXX.XXX.2.46:5060 -> XXX.XXX.2.40:5060 > BYE sip:user@domain.net:8923 SIP/2.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK3fb57de2;rport. > Route: > <sip:XXX.XXX.2.40;transport=tcp;ftag=3cc4119df7fb42e9a4bccbb508466931;lr=on>. > From: <sip:55555@domain.net>;tag=as3338df1d. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=3cc4119df7fb42e9a4bccbb508466931;epid=f272a56bbd. > Contact: <sip:55555@XXX.XXX.2.46>. > Call-ID: 3ff00aceaba0435a9ed23dc01dc7a77a. > CSeq: 102 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > > > > T 2006/03/02 15:07:59.643864 XXX.XXX.2.40:42664 -> XXX.XXX.2.40:5060 [AP] > BYE > sip:XXX.XXX.2.40;transport=tcp;ftag=3cc4119df7fb42e9a4bccbb508466931;lr=on > SIP/2.0. > Record-Route: <sip:XXX.XXX.2.40;ftag=as3338df1d;lr=on>. > Via: SIP/2.0/TCP XXX.XXX.2.40;branch=z9hG4bK5419.1af2f513.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK3fb57de2;rport=5060. > From: <sip:55555@domain.net>;tag=as3338df1d. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=3cc4119df7fb42e9a4bccbb508466931;epid=f272a56bbd. > Contact: <sip:55555@XXX.XXX.2.46>. > Call-ID: 3ff00aceaba0435a9ed23dc01dc7a77a. > CSeq: 102 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 69. > Content-Length: 0. > T 2006/03/02 15:07:59.656229 XXX.XXX.2.40:5060 -> XXX.XXX.2.40:42664 [AP] > SIP/2.0 404 User Not Found. > Via: SIP/2.0/TCP XXX.XXX.2.40;branch=z9hG4bK5419.1af2f513.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK3fb57de2;rport=5060. > From: <sip:55555@domain.net>;tag=as3338df1d. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=3cc4119df7fb42e9a4bccbb508466931;epid=f272a56bbd. > Call-ID: 3ff00aceaba0435a9ed23dc01dc7a77a. > CSeq: 102 BYE. > Server: OpenSer (1.0.1-tls (i386/linux)). > Content-Length: 0.

2 1

[Users] how to route a call between servers
by vikram sarathy 02 Mar '06

02 Mar '06

hi, i have 3 servers in my setup, in which each server has a user registered. i need to route a call from one server to other. in first case i kept all servers in same domine and later i need to check with all servers in different domains. can any one tell me is there any way to achive this. with regards vicky vicky

2 1

← Newer
1
...
44
45
46
47
48
49
50
51
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users March 2006