sr-users August 2004

sr-users@lists.kamailio.org

181 participants
424 discussions

[Serusers] Nathelper with rtpproxy or mediaproxy?

by Michael Shuler

They both look to me like they do the same thing, which is better? ---------------------------------------- Michael Shuler, C.E.O. BitWise Systems, Inc. 682 High Point Lane East Peoria, IL 61611 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: mike(a)bwsys.net Customer Service: (877) 976-0711

19 years, 8 months

FW: [Serusers] Sort question regarding INVITE and timeout

by Vitaly Nikolaev

Cisco 5350 There is timers but maxim is 1000 milisec I increase all of them and have now: sip-ua nat symmetric role active nat symmetric check-media-src timers trying 1000 timers expires 360000 timers connect 1000 timers rel1xx 1000 sip-server dns:XXX I guess that more correct way to do that ON ser.. to limit amount of RE INVITE before SER decide that that call fail Is it possible ? > -----Original Message----- > From: mark wehberg [mailto:mark.wehberg@clearviewcatv.net] > Sent: Wednesday, August 25, 2004 3:20 PM > To: Vitaly Nikolaev > Subject: RE: [Serusers] Sort question regarding INVITE and timeout > > There should be timers that you can set on the cisco, which cisco are u > using? > > -----Original Message----- > From: Vitaly Nikolaev [mailto:vitaly@switchgate.com] > Sent: Wednesday, August 25, 2004 3:18 PM > To: mark wehberg > Cc: serusers(a)lists.iptel.org > Subject: RE: [Serusers] Sort question regarding INVITE and timeout > > > Well, is it working in case when call coming from Cisco GW and > client that supposed to get call offline but still registered on SER ? > > In this case cisco send cancel to SER before fr_inv_timer triggered I > thing 3-4 reinvite without answer is enough to realize that client is > offline and give up with it but how to program it > > > > > > -----Original Message----- > > From: mark wehberg [mailto:mark.wehberg@clearviewcatv.net] > > Sent: Wednesday, August 25, 2004 3:14 PM > > To: Vitaly Nikolaev > > Subject: RE: [Serusers] Sort question regarding INVITE and timeout > > > > I have not used SER's voicemail server, but have sent calls to other > > voicemail servers under failure conditions.... > > > > -Mark > > > > > > > > Set your timer > > > > modparam("tm", "fr_inv_timer", 20 ) > > > > > > then set the flag before the t_relay > > > > t_on_failure("1"); > > t_relay_to_udp("xxx.xxx.xxx.xxx", "5060"); > > > > > > then set up your failure case > > > > > > failure_route[1] { > > log(1,"failed Call"); > > t_relay_to_udp("xxx.xxx.xxx", "5060"); > > } > > > > > > -----Original Message----- > > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] > On > > Behalf Of Vitaly Nikolaev > > Sent: Wednesday, August 25, 2004 2:40 PM > > To: serusers(a)lists.iptel.org > > Subject: [Serusers] Sort question regarding INVITE and timeout > > > > Hello, > > > > I could not find answer to my question in the list though it was > > discussed before without ultimate solution :) > > > > > > I have customer who registered on SER then go offline (internet > > connecting or power problem) > > > > Call coming from cisco and going via SER to this customers. > > > > I have INVITE from cisco > > Ser sends trying to cisco > > > > Ser send invite to customer > > Ser send invite to customer > > Ser send invite to customer > > Ser send invite to customer > > > > > > Cisco sends cancel to ser... > > > > > > And that is it. And that is not what I want... because falure route > > supposed to get control over the call and send it to voicemail. I > tried > > playing with timers on cisco/ser but without success. > > > > Any ideas? > > > > Thank you > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > >

19 years, 8 months

[Serusers] Is there an interface to PostGreSQL instead of mySQL?

by John Draper

Hi, Is anyone working on a PostGreSQL interface to SER? If so, who? John

19 years, 8 months

[Serusers] NAT

by Michael Shuler

I'm trying to understand the NAT process, can someone tell me if this is correct Here is my original INVITE from client IP 123.123.123.123 which is received from port 5002 (because its behind a NAT router): INVITE sip:2175850357@sip2.bwsys.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.2:5060;branch=z9hG4bK-4qxhdyvuyz6t;rport From: "Mike's Springfield snom 1" <sip:MikeSnom1@sip2.bwsys.net>;tag=k91lt5ydem To: <sip:2175850357@sip2.bwsys.net;user=phone> Call-ID: 3c275253ad55-krb7eduyhd2m@192-168-100-2 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:MikeSnom1@192.168.100.2:5060;line=rygvn8mw> P-Key-Flags: keys="3" User-Agent: snom200-3.42 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 256 v=0 o=root 20147217 20147217 IN IP4 192.168.100.2 s=call c=IN IP4 192.168.100.2 t=0 0 m=audio 10048 RTP/AVP 18 0 8 101 a=rtpmap:18 g729/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Now this invite is going to get forwarded to my TNT (with a public IP) which has SIP capabilities and hands of PRI to a phone switch. So to make NAT support work what would this message look like after being proxied and NAT "adjusted" and what function in SER would be needed to make the changes happen? Now I know the proxy should be able to learn the port that the INVITE comes in on and can send SIP messages back via the learned port, but I do not understand how the RTP stream works when NAT is involved. My guess is that anything what says 192.168.100.2 (the NAT'ed IP) needs to get changed to the proxy's public IP of 198.88.216.87:5060. This would include the Via and Contact. Now the SDP "c" line has to be changed because it says 192.168.100.2 but it should be changed to 123.123.123.123 (the clients learned IP). Now that leaves us to the problem of the RTP port. The client says it wants to receive on 10048, but how does the generic NAT router know that the snom wants this port opened? ---------------------------------------- Michael Shuler, C.E.O. BitWise Systems, Inc. 682 High Point Lane East Peoria, IL 61611 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: mike(a)bwsys.net Customer Service: (877) 976-0711

19 years, 8 months

[Serusers] ERROR: bye (AmRequest.cpp:310): AmRequest::bye: 481 Call Leg/Transaction Does Not Exist

by Rafael J. Risco G.V

Hello In my ser implementation I have radius accounting for missed calls (busy, not response or offline) and also sems voicemail for those missed calls. My problem is that 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 (just an start record without stop) then I see this error in sems debug after 30 seconds: ERROR: bye (AmRequest.cpp:310): AmRequest::bye: 481 Call Leg/Transaction Does Not Exist please see the ngrep info I send below... some idea? Also I need to know if its possible to do this: 1. announcement "user_is_offline.wav" if the user exist but it is not connected and it is not in voicemail group. 2. announcement "user_not_exist.wav" if the user is not in the subscriber table. can someone share their configuration file to do this? Rafael PS: ngrep -v REGISTER port 5060 below and my configs files in http://rrisco.pub.millicom.com.pe/ser_config_files.txt # U 10.0.0.236:5060 -> 200.110.2.131:5060 INVITE sip:6604000@call.millicom.com.pe SIP/2.0..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc500bed0a4387..From: <sip:66030 ;tag=c500bed0a4..To">00(a)call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@ 10.0.0.236..CSeq: 387 INVITE..Supported: timer..Min-SE: 1800..Date: Thu, 30 Apr 1970 21:29:41 GMT..User-Agent: AddPac SIP Gatew ay..Contact: sip:6603000@10.0.0.236..Content-Type: application/sdp..Content-Length: 182..Max-Forwards: 70....v=0..o=- 10358981 10358981 IN IP4 10.0.0.236..s=AddPac Gateway SDP..c=IN IP4 10.0.0.236..t=0 0..m=audio 23888 RTP/AVP 4 18 0 101..a=rtpmap:101 te lephone-event/8000..a=fmtp:101 0-15.. # U 200.110.2.131:5060 -> 10.0.0.236:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc500bed0a4387..From: <sip:66 ;tag=c500bed0a4..To">03000(a)call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055 b2@10.0.0.236..CSeq: 387 INVITE..Server: Sip EXpress router (0.8.99-dev (i386/linux))..Content-Length: 0..Warning: 392 200.110. 2.131:5060 "Noisy feedback tells: pid=1897 req_src_ip=10.0.0.236 req_src_port=5060 in_uri=sip:6604000@call.millicom.com.pe out _uri=sip:6604000@172.18.96.254 via_cnt==1".... # U 200.110.2.131:5060 -> 172.18.96.254:5060 INVITE sip:6604000@172.18.96.254 SIP/2.0..Record-Route: <sip:200.110.2.131;ftag=c500bed0a4;lr=on>..Via: SIP/2.0/UDP 200.110.2.1 31;branch=z9hG4bK2d89.41143714.0..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc500bed0a4387..From: <sip:6603000@call.millico m.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSe q: 387 INVITE..Supported: timer..Min-SE: 1800..Date: Thu, 30 Apr 1970 21:29:41 GMT..User-Agent: AddPac SIP Gateway..Contact: si p:6603000@10.0.0.236..Content-Type: application/sdp..Content-Length: 182..Max-Forwards: 16....v=0..o=- 10358981 10358981 IN IP4 10.0.0.236..s=AddPac Gateway SDP..c=IN IP4 10.0.0.236..t=0 0..m=audio 23888 RTP/AVP 4 18 0 101..a=rtpmap:101 telephone-event/8 000..a=fmtp:101 0-15.. # U 172.18.96.254:5060 -> 200.110.2.131:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 200.110.2.131;branch=z9hG4bK2d89.41143714.0..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4 bKc500bed0a4387..From: <sip:6603000@call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>..Call-ID: c510 9e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSeq: 387 INVITE..User-Agent: AddPac SIP Gateway..Content-Length: 0.... # U 172.18.96.254:5060 -> 200.110.2.131:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 200.110.2.131;branch=z9hG4bK2d89.41143714.0..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG 4bKc500bed0a4387..From: <sip:6603000@call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>;tag=4a40522ea 4..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSeq: 387 INVITE..User-Agent: AddPac SIP Gateway..Contact: sip:660 4000@172.18.96.254..Content-Length: 0..Record-Route: <sip:200.110.2.131;ftag=c500bed0a4;lr=on>.... # U 200.110.2.131:5060 -> 10.0.0.236:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc500bed0a4387..From: <sip:6603000@call.millicom.com.pe>;tag =c500bed0a4..To: <sip:6604000@call.millicom.com.pe>;tag=4a40522ea4..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..C Seq: 387 INVITE..User-Agent: AddPac SIP Gateway..Contact: sip:6604000@172.18.96.254..Content-Length: 0..Record-Route: <sip:200. 110.2.131;ftag=c500bed0a4;lr=on>.... ######### U 10.0.0.236:5060 -> 200.110.2.131:5060 CANCEL sip:6604000@call.millicom.com.pe SIP/2.0..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc500bed0a4387..From: <sip:66030 ;tag=c500bed0a4..To">00(a)call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@ 10.0.0.236..CSeq: 387 CANCEL..Date: Thu, 30 Apr 1970 21:29:43 GMT..User-Agent: AddPac SIP Gateway..Content-Length: 0..Max-Forwa rds: 70.... # U 200.110.2.131:5060 -> 172.18.96.254:5060 CANCEL sip:6604000@172.18.96.254 SIP/2.0..Record-Route: <sip:200.110.2.131;ftag=c500bed0a4;lr=on>..Via: SIP/2.0/UDP 200.110.2.1 31;branch=z9hG4bK2d89.41143714.0..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc500bed0a4387..From: <sip:6603000@call.millico m.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSe q: 387 CANCEL..Date: Thu, 30 Apr 1970 21:29:43 GMT..User-Agent: AddPac SIP Gateway..Content-Length: 0..Max-Forwards: 16.... # U 200.110.2.131:5060 -> 10.0.0.236:5060 SIP/2.0 200 cancelling..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc500bed0a4387..From: <sip:6603000@call.millicom.com.pe>; tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-4925..Call-ID: c5109e00-cec1-be2d-8 4d0-0002a40055b2@10.0.0.236..CSeq: 387 CANCEL..Server: Sip EXpress router (0.8.99-dev (i386/linux))..Content-Length: 0..Warning : 392 200.110.2.131:5060 "Noisy feedback tells: pid=1898 req_src_ip=10.0.0.236 req_src_port=5060 in_uri=sip:6604000@call.milli com.com.pe out_uri=sip:6604000@172.18.96.254 via_cnt==1".... ### U 172.18.96.254:5060 -> 200.110.2.131:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 200.110.2.131;branch=z9hG4bK2d89.41143714.0..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc5 00bed0a4387..From: <sip:6603000@call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>..Call-ID: c5109e00 -cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSeq: 387 CANCEL..User-Agent: AddPac SIP Gateway..Content-Length: 0..Record-Route: <si p:200.110.2.131;ftag=c500bed0a4;lr=on>.... ### U 172.18.96.254:5060 -> 200.110.2.131:5060 SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 200.110.2.131;branch=z9hG4bK2d89.41143714.0..Via: SIP/2.0/UDP 10.0.0.236:5060; branch=z9hG4bKc500bed0a4387..From: <sip:6603000@call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>;ta g=4a40522ea4..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSeq: 387 INVITE..User-Agent: AddPac SIP Gateway..Conte nt-Length: 0.... # U 200.110.2.131:5060 -> 172.18.96.254:5060 ACK sip:6604000@172.18.96.254 SIP/2.0..Via: SIP/2.0/UDP 200.110.2.131;branch=z9hG4bK2d89.41143714.0..From: <sip:6603000@call.mi llicom.com.pe>;tag=c500bed0a4..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..To: <sip:6604000@call.millicom.com.pe> ;tag=4a40522ea4..CSeq: 387 ACK..User-Agent: Sip EXpress router(0.8.99-dev (i386/linux))..Content-Length: 0.... # U 200.110.2.131:5060 -> 10.0.0.236:5060 SIP/2.0 180 ringing..Record-Route: <sip:200.110.2.131;ftag=c500bed0a4;lr=on>..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc5 00bed0a4387..From: <sip:6603000@call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>;tag=0000356755B227 CF..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSeq: 387 INVITE..Contact: <sip:6604000@200.110.2.131:5090>..Serv er: Sip EXpress router (0.8.99-dev (i386/linux))..Content-Length: 0..Warning: 392 200.110.2.131:5090 "Noisy feedback tells: pi d=1933 req_src_ip=200.110.2.131 req_src_port=5060 in_uri=sip:6604000@call.millicom.com.pe:5090 out_uri=sip:6604000@call.millico m.com.pe:5090 via_cnt==0".... # . . . . # U 200.110.2.131:5060 -> 10.0.0.236:5060 SIP/2.0 200 OK..Record-Route: <sip:200.110.2.131;ftag=c500bed0a4;lr=on>..Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKc500bed 0a4387..From: <sip:6603000@call.millicom.com.pe>;tag=c500bed0a4..To: <sip:6604000@call.millicom.com.pe>;tag=0000356755B227CF..C all-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSeq: 387 INVITE..Contact: <sip:6604000@200.110.2.131:5090>..Content-T ype: application/sdp..Server: Sip EXpress router (0.8.99-dev (i386/linux))..Content-Length: 123..Warning: 392 200.110.2.131:509 0 "Noisy feedback tells: pid=1933 req_src_ip=200.110.2.131 req_src_port=5060 in_uri=sip:6604000@call.millicom.com.pe:5090 out_ uri=sip:6604000@call.millicom.com.pe:5090 via_cnt==0"....v=0..o=username 0 0 IN IP4 200.110.2.131..s=session..c=IN IP4 200.110. 2.131..t=0 0..m=audio 1418 RTP/AVP 0..a=rtpmap:0 /... # U 200.110.2.131:5060 -> 10.0.0.236:5060 BYE sip:6603000@10.0.0.236 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200.110.2.131;ftag=0000356755B227CF;lr=on>..Via: SIP/2 .0/UDP 200.110.2.131;branch=z9hG4bK0e89.94661d82.0..Via: SIP/2.0/UDP 200.110.2.131:5090;branch=z9hG4bK0e89.ce855942.0..To: <sip :6603000@call.millicom.com.pe>;tag=c500bed0a4..From: <sip:6604000@call.millicom.com.pe>;tag=0000356755B227CF..CSeq: 388 BYE..Ca ll-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..Content-Length: 0..User-Agent: Sip EXpress router(0.8.99-dev (i386/linu x))..Contact: <sip:6604000@200.110.2.131:5090>..P-MsgFlags: 0..Min-SE: 1800..Date: Thu, 30 Apr 1970 21:29:41 GMT.... # U 10.0.0.236:5060 -> 200.110.2.131:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist..Via: SIP/2.0/UDP 200.110.2.131;branch=z9hG4bK0e89.94661d82.0..Via: SIP/2.0/UDP 200.110.2.131:5090;branch=z9hG4bK0e89.ce855942.0..From: <sip:6604000@call.millicom.com.pe>;tag=0000356755B227CF..To: <sip:6603 ;tag=c500bed0a4..Call-ID">000(a)call.millicom.com.pe>;tag=c500bed0a4..Call-ID: c5109e00-cec1-be2d-84d0-0002a40055b2@10.0.0.236..CSeq: 388 BYE..User-Agent: AddPac SIP Gateway..Content-Length: 0.... ######################### --------------------------------- Do You Yahoo!? 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19 years, 8 months

RE: [Serusers] Sort question regarding INVITE and timeout

by Vitaly Nikolaev

Well, is it working in case when call coming from Cisco GW and client that supposed to get call offline but still registered on SER ? In this case cisco send cancel to SER before fr_inv_timer triggered I thing 3-4 reinvite without answer is enough to realize that client is offline and give up with it but how to program it > -----Original Message----- > From: mark wehberg [mailto:mark.wehberg@clearviewcatv.net] > Sent: Wednesday, August 25, 2004 3:14 PM > To: Vitaly Nikolaev > Subject: RE: [Serusers] Sort question regarding INVITE and timeout > > I have not used SER's voicemail server, but have sent calls to other > voicemail servers under failure conditions.... > > -Mark > > > > Set your timer > > modparam("tm", "fr_inv_timer", 20 ) > > > then set the flag before the t_relay > > t_on_failure("1"); > t_relay_to_udp("xxx.xxx.xxx.xxx", "5060"); > > > then set up your failure case > > > failure_route[1] { > log(1,"failed Call"); > t_relay_to_udp("xxx.xxx.xxx", "5060"); > } > > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On > Behalf Of Vitaly Nikolaev > Sent: Wednesday, August 25, 2004 2:40 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Sort question regarding INVITE and timeout > > Hello, > > I could not find answer to my question in the list though it was > discussed before without ultimate solution :) > > > I have customer who registered on SER then go offline (internet > connecting or power problem) > > Call coming from cisco and going via SER to this customers. > > I have INVITE from cisco > Ser sends trying to cisco > > Ser send invite to customer > Ser send invite to customer > Ser send invite to customer > Ser send invite to customer > > > Cisco sends cancel to ser... > > > And that is it. And that is not what I want... because falure route > supposed to get control over the call and send it to voicemail. I tried > playing with timers on cisco/ser but without success. > > Any ideas? > > Thank you > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

19 years, 8 months

[Serusers] Can't connect to local MySQL server

by Pali Majernik

Hi, I installed SER exactly following INSTALL guide. I am running on the same machine MySQL 4.0.20 and I successfuly initialised ser database. After running ser by "serctl start" I got message: Starting SER : cat: /var/run/ser.pid: No such file or directory started pid () ~# Of course ser is not running. In syslog I can see messages: ser: WARNING: could not rev. resolve 10.0.0.8 /usr/sbin/ser[579]: connect_db(): Can't connect to local MySQL server \ through socket '/var/run/mysqld/mysqld.sock' (2) /usr/sbin/ser[579]: db_init(): Error while trying to connect database /usr/sbin/ser[579]: mod_init(): Error while connecting database /usr/sbin/ser[579]: init_mod(): Error while initializing module usrloc The IP 10.0.0.8 is address of PC where I'm running this SER and MySQL. Any hint/idea? Pali

19 years, 8 months

[Serusers] Problem Install

by Shiran Guez

Hi I have downloaded ser-0.8.14_src.tar and did tar zxvf ser-0.8.14_src.tar.gz make all make prefix=/ install #and then export SIP_DOMAIN="sipser.zeurolink.co.il" # then I tried to start the server serctl start Starting SER : cat: /var/run/ser.pid: No such file or directory started pid() please help Shiran Guez

19 years, 8 months

[Serusers] Question on sms module

by Lakmal Silva

Hi guys, I'm trying to implement the sms functionality. I already have a sms gateway on a seperate server than ser is running. What I want to do is to send the sms messages via this server. Is this possible? (In a similar way we route calls to a PSTN gateway) According to the doc what I realised it that the modem needs to be connected to the machine where ser is running. The other question is what sort of a client can be used for sending sms? -- Regards, Lakmal Lankacom Services (Pvt) Ltd. 65C, Dharmapala Mawatha, Colombo 07. Sri Lanka. Tel: +94-11-2437545 www.lankacom.net

19 years, 8 months

[Serusers] Window Messenger + RTP Proxy

by Ram Sujay

Hi, I am trying to establish call b/w window messenger (5.0.0482) and X-lite. i am able to place call but communication is one way means voice is transmitted from X-lite to messenger but voice from messenger is not coming. media is established through rtpproxy... is window messenger support symmetric client same problem if both client is window messenger. I think , it is NAT problem. but how to fix the problem can u help version of ser -0.8.13 version nathelper- v 1.51 rtpproxy version - v 1.19 plzz can anyone tell me whats the problem Thanks Ram _______________________________ Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now. http://promotions.yahoo.com/goldrush

19 years, 8 months

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sr-users August 2004