sr-users July 2005

sr-users@lists.kamailio.org

199 participants
544 discussions

[Serusers] Over-long fields can crash SER with PostgreSQL
by Karl A. Krueger 28 Jul '05

28 Jul '05

We are using SER with the PostgreSQL module, so that SER and our Asterisk PSTN gateway and voice mail system can use the same back-end database for user records. (Anyone who's interested in how we did this, let me know -- it's still a little shaky, but it pretty much works.) Recently we have encountered a problem: it seems that if any database error occurs, SER hangs or crashes. We have so far been able to mostly work around these conditions by being Really Careful with the database, but it is a little unsettling. The database schema we're working from is practically identical to the MySQL schema that ships with SER. Notably for this problem the lengths of all CHARACTER VARYING fields are the same. The User-Agent field is CHARACTER VARYING(50), for instance. So ... today boss got a copy of Xten's "eyeBeam" software, set it up to register to our SER ... and SER promptly locked up. I looked in the PostgreSQL database log and found this: Feb 10 13:39:32 mabell postgres[13058]: [267] ERROR: value too long for type character varying(50) Whoops. SER tried to insert eyeBeam's User-Agent data, PostgreSQL returned a data integrity error, and SER went bye-bye. So I did a little ALTER TABLE and adjusted the field to length 128 ... then put this in ser.cfg: # Disallow long user-agent if (search("^User-Agent: .{100,}") ) { log(1, "LOG: User agent too long\n"); sl_send_reply("479", "User agent too long!"); break; }; But this is obviously a kludge ... and doesn't fix the problem of one of the *other* fields being over-long. Anyone have any thoughts on this? Robustness patches for the PostgreSQL module or usrloc? Better ser.cfg workarounds to protect the world from malformed or over-long SIP header fields? Should I just change all the database fields to TEXT instead of CHARACTER VARYING(whatever) ? -- Karl A. Krueger <kkrueger(a)whoi.edu> Network Security -- Linux/Unix Systems Support -- Etc. Woods Hole Oceanographic Institution

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Re: [Serusers] asterisk and ser combine, 407 proxy authentucation required !
by Iqbal 28 Jul '05

28 Jul '05

tell ser that all packets from asterisk IP dont need to be authenticated, use trusted_ip setup for it, that should do the trick Iqbal On 7/28/2005, "zhaomin" <zhaomin(a)kingdream.com> wrote: >hello! > I want to realized asterisk+ser combine working,ser is working fine and asterisk working well I think,I configure sip.conf and extensions.conf voicemail.conf as the asterisk handbook ,but when I registered 2000 to dial 2001 but I get error :407 proxy authentucation required, > where is problem in asterisk or ser ? >blow is the log of x-lite diagnostic,Very appreciation anybody help! >SEND TIME: 201758109 >SEND >> 221.202.129.167:5060 >INVITE sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 INVITE >Max-Forwards: 70 >Content-Type: application/sdp >User-Agent: X-Lite release 1103m >Content-Length: 244 > >v=0 >o=2000 201757640 201758109 IN IP4 10.1.16.34 >s=X-Lite >c=IN IP4 10.1.16.34 >t=0 0 >m=audio 8000 RTP/AVP 0 98 97 101 >a=rtpmap:0 pcmu/8000 >a=rtpmap:98 iLBC/8000 >a=rtpmap:97 speex/8000 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-15 > >RECEIVE TIME: 201758859 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 407 Proxy Authentication Required >Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 INVITE >Proxy-Authenticate: Digest realm="netvoip.cn", nonce="42e896c750edabc9057e78d345379a4991b02378" >Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) >Content-Length: 0 >Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28404 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@netvoip.cn via_cnt==1" > > >SEND TIME: 201758859 >SEND >> 221.202.129.167:5060 >ACK sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 ACK >Max-Forwards: 70 >Content-Length: 0 > > >SEND TIME: 201758906 >SEND >> 221.202.129.167:5060 >INVITE sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >Proxy-Authorization: Digest username="2000",realm="netvoip.cn",nonce="42e896c750edabc9057e78d345379a4991b02378",response="337cb9deeceb1bf01e9768b6a5452568",uri="sip:2001@netvoip.cn" >Max-Forwards: 70 >Content-Type: application/sdp >User-Agent: X-Lite release 1103m >Content-Length: 244 > >v=0 >o=2000 201757640 201758109 IN IP4 10.1.16.34 >s=X-Lite >c=IN IP4 10.1.16.34 >t=0 0 >m=audio 8000 RTP/AVP 0 98 97 101 >a=rtpmap:0 pcmu/8000 >a=rtpmap:98 iLBC/8000 >a=rtpmap:97 speex/8000 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-15 > >RECEIVE TIME: 201759640 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 100 trying -- your call is important to us >Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) >Content-Length: 0 >Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28398 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@221.202.129.167:5061 via_cnt==1" > > >RECEIVE TIME: 201759671 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 407 Proxy Authentication Required >Via: SIP/2.0/UDP 10.1.16.34:5060;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=as6e0c8cea >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >User-Agent: Asterisk PBX >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >Contact: <sip:2001@221.202.129.167:5061> >Proxy-Authenticate: Digest realm="asterisk", nonce="43fb4632" >Content-Length: 0 > > >SEND TIME: 201759671 >SEND >> 221.202.129.167:5060 >ACK sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=as6e0c8cea >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 ACK >Max-Forwards: 70 >Content-Length: 0 > >

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[Serusers] SER Asterisk Voicemail - Message Playback
by Aisling 28 Jul '05

28 Jul '05

Hi Everyone, Unfortunately when I was upgrading my version of SER to 0.9.0 I deleted my config file which contained code for forwarding to asterisk voicemail.Therefore I am attempting to set up all over again (Doh! I know!) :-( Anyhow SER seems to be correctly forwarding to Asterisk, the user can leave a message if the phone isn't picked up after 3 rings and then the message is sent to the users email account. I want the user to also be able to listen to their voicemails from the end user device. It is my understanding from what I have coded, that user should be able to dial 9999 and when SER receives this it sends it to Asterisk' VoiceMailMain where the user will be asked for an account name and password. However there are problems when the user dials 9999. The user can hear nothing on the phone. Errors appear on both the SER screen and Asterisk screen so I am not sure what the problem is: On SER it says: Warning: sl_send_reply: I won't send a reply for ACK!! Forwarding to asterisk14(27355) contact_parser(): Empty body Parse_contact(): Error while parsing Get_contact_uri: Error while parsing Contact body On Asterisk it says: -- Executing VoiceMailMain("SIP/2092-dd89", "2092") in new stack -- Playing 'vm-login' (language 'en') WARNING: app_voicemail.c:3333 vm_execmain: Couldn't read username Spawn extension (test, 9999, 1) exited non-zero on 'SIP/2092-c660' The relevant parts of my ser.cfg and asterisk config files are shown below. Many thanks, Aisling. ------------------------------------------------------------------------ ----------------------------------- ser.cfg ------------------------------------------------------------------------ ---------------------------------- #Call Type Processing Section if(uri==myself){ if(uri=~"^sip:9[0-9]*@serveraddress"){ log(1, "forwarding to asterisk"); rewritehostport("serveraddress:port"); append_branch(); t_relay_to_udp("serveraddress", "port"); break; } if (method == "INVITE"){ t_on_failure("1"); route(3); break; } failure_route[1]{ revert_uri(); rewritehostport("serveraddress:port"); append_branch(); t_relay_to_udp("serveraddress", "port"); break(); } ------------------------------------------------------------------------ -------------------------------------- sip.conf ------------------------------------------------------------------------ ------------------------------------- [general] context=test port=5062 bindaddr=0.0.0.0 srvlookup=yes [2092] type=friend username=2092 canreinvite=no context=test mailbox=2092 host=dynamic nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw [314] type=friend username=314 canreinvite=no context=test mailbox=314 host=dynamic nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw ------------------------------------------------------------------------ ----------------------------------- extensions.conf ------------------------------------------------------------------------ ---------------------------------- [test] ;leave voice messages exten => 2092, 1, Voicemail(u2092) exten => 2092, 2, Hangup exten => 314, 1, Voicemail(u314) exten => 314, 2, Hangup ;play voice messages exten => 9999, 1, VoiceMailMain, 2092 exten => 9999, 2, Hangup exten => 9999, 1, VoiceMailMain, 314 exten => 9999, 2, Hangup ------------------------------------------------------------------------ ------------------------------------- voicemail.conf ------------------------------------------------------------------------ -------------------------------------- [default] 2092 => 1234, 2092, emailaddress 314 => 1234, 314, emailaddress -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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RE : RE : [Serusers] forwarding and accounting
by Olivier Taylor 28 Jul '05

28 Jul '05

Sorry guy, u think I don't contribute, that's ok, but I try helping as much as I can. Respect for the job done by the Ser team, but sorry, I and many of us don't understand why we have now two Ser...(an open one and an open one...) If u really want to contribute, don't fork, just contribute, otherwise it will be another software. If you don't want people use your work, go and join the Microsoft team and forget free software. That's only the poor opinion of a guy using unfairly free software. Olivier Ps: make ur opinion available for all and put serusers in copy for the next mail. -----Message d'origine----- De : Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Envoyé : jeudi 28 juillet 2005 18:28 À : Olivier Taylor Objet : Re: RE : [Serusers] forwarding and accounting hmmm... I really do find annoying and unfair this kind of judgement from somebody who just enjoyed for free others people work without any contribution.....I will kindly ask you consider this before expressing such opinions. regards, bogdan Olivier Taylor wrote: >I can't accept the fork with Openser, sorry, I will find another >solution. Openser just have to rejoin Ser, that forking is very >annoying and probably unfair. > >Olivier > >-----Message d'origine----- >De : Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Envoyé : jeudi 28 juillet 2005 17:57 >À : Olivier Taylor >Objet : Re: [Serusers] forwarding and accounting > > >Hi Olivier, > >what you are looking for is multi-leg call accounting - SER has no >support for it, but OpenSER has. >see http://openser.org/pipermail/users/2005-June/000140.html > >regards, >bogdan > >Olivier Taylor wrote: > > > >>hi all, >> >>what about the accounting when forwarding a call. >> >>A-->B--ForwardTo-->C >> >>Forwarding is OK, even to Pstn, but... >>B as to pay the forwarding, but i don't see any record on acc neither >>radacct concerning this call for B. >> >>Any Idea? >> >>Olivier >> >>---------------------------------------------------------------------- >>- >>- >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> > > >

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[Serusers] forwarding and accounting
by Olivier Taylor 28 Jul '05

28 Jul '05

hi all, what about the accounting when forwarding a call. A-->B--ForwardTo-->C Forwarding is OK, even to Pstn, but... B as to pay the forwarding, but i don't see any record on acc neither radacct concerning this call for B. Any Idea? Olivier

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[Users] timer question
by Klaus Darilion 28 Jul '05

28 Jul '05

Hi! What should happen if the invite_timer expires and there is no failure route defined? In my case happens nothing, the phone is still ringing. Shouldn't openser (CVS) cancel the call and responds with 408? If I define a failure_route, everything works fine. regards, klaus debuglog: 9(21752) DEBUG: reply relayed. buf=0x81148d0: SIP/2.0 1..., shmem=0x40606568: SIP/2.0 1 9(21752) reply_received: FR_INV_TIMER = 10 9(21752) DEBUG: add_to_tail_of_timer[1]: 0x40604d7c 9(21752) DEBUG:destroy_avp_list: destroying list (nil) 9(21752) receive_msg: cleaning up after 10 seconds: 10(21756) DEBUG: timer routine:1,tl=0x40604d7c next=(nil) 10(21756) DEBUG: final_response_handler: transaction silently dropped (0x40604c50) 10(21756) DEBUG: add_to_tail_of_timer[2]: 0x40604c98 phone is still ringing, some seconds later: 10(21756) DEBUG: timer routine:2,tl=0x40604c98 next=(nil) 10(21756) DEBUG: wait_handler : removing 0x40604c50 from table 10(21756) DEBUG: delete transaction 0x40604c50 10(21756) DEBUG: wait_handler : done phone is still ringing :-(

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[Serusers] Presence and messaging: UA with SER
by Cameron Beattie 28 Jul '05

28 Jul '05

I want to be able to offer the following functionality (in addition to calling) to my users namely: notification when a user is on/offline, directory search, and chat. These simple requirements are met with ease by Skype. And yet there doesn't seem to be any combination of SER and a freely-available Windows user agent that offers this. I have been a little bit out of touch with this recently so things may have moved on since I last checked. The main problem seems to be on the UA side. I've tried (amongst others) X-Ten, SJPhone, PhoneGaim, Firefly, and Windows Messenger - none of them come close to Skype's ease of configuration and use. Am I missing something? Can anyone offer some suggestions about how to offer this functionality? Regards Cameron

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[Serusers] asterisk and ser combine, 407 proxy authentucation required !
by zhaomin 28 Jul '05

28 Jul '05

hello! I want to realized asterisk+ser combine working,ser is working fine and asterisk working well I think,I configure sip.conf and extensions.conf voicemail.conf as the asterisk handbook ,but when I registered 2000 to dial 2001 but I get error :407 proxy authentucation required, where is problem in asterisk or ser ? blow is the log of x-lite diagnostic,Very appreciation anybody help! SEND TIME: 201758109 SEND >> 221.202.129.167:5060 INVITE sip:2001@netvoip.cn SIP/2.0 Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn> Contact: <sip:2000@10.1.16.34:5060> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53903 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 244 v=0 o=2000 201757640 201758109 IN IP4 10.1.16.34 s=X-Lite c=IN IP4 10.1.16.34 t=0 0 m=audio 8000 RTP/AVP 0 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 201758859 RECEIVE << 221.202.129.167:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53903 INVITE Proxy-Authenticate: Digest realm="netvoip.cn", nonce="42e896c750edabc9057e78d345379a4991b02378" Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) Content-Length: 0 Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28404 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@netvoip.cn via_cnt==1" SEND TIME: 201758859 SEND >> 221.202.129.167:5060 ACK sip:2001@netvoip.cn SIP/2.0 Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b Contact: <sip:2000@10.1.16.34:5060> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53903 ACK Max-Forwards: 70 Content-Length: 0 SEND TIME: 201758906 SEND >> 221.202.129.167:5060 INVITE sip:2001@netvoip.cn SIP/2.0 Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn> Contact: <sip:2000@10.1.16.34:5060> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53904 INVITE Proxy-Authorization: Digest username="2000",realm="netvoip.cn",nonce="42e896c750edabc9057e78d345379a4991b02378",response="337cb9deeceb1bf01e9768b6a5452568",uri="sip:2001@netvoip.cn" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 244 v=0 o=2000 201757640 201758109 IN IP4 10.1.16.34 s=X-Lite c=IN IP4 10.1.16.34 t=0 0 m=audio 8000 RTP/AVP 0 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 201759640 RECEIVE << 221.202.129.167:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53904 INVITE Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) Content-Length: 0 Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28398 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@221.202.129.167:5061 via_cnt==1" RECEIVE TIME: 201759671 RECEIVE << 221.202.129.167:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.16.34:5060;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn>;tag=as6e0c8cea Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53904 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2001@221.202.129.167:5061> Proxy-Authenticate: Digest realm="asterisk", nonce="43fb4632" Content-Length: 0 SEND TIME: 201759671 SEND >> 221.202.129.167:5060 ACK sip:2001@netvoip.cn SIP/2.0 Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 To: <sip:2001@netvoip.cn>;tag=as6e0c8cea Contact: <sip:2000@10.1.16.34:5060> Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 CSeq: 53904 ACK Max-Forwards: 70 Content-Length: 0

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[Users] creating postgres tables
by Klaus Darilion 28 Jul '05

28 Jul '05

Hi! I recently checked in scripts/postgresqldb.sh - a rewrite of the mysql version for postgresql. Currently tested are only the "create" and "drop" commands, thus the other commands are disabled. I had to modify the tables defintions and I'm not sure if this broke something. Thus, please report problems. regards, klaus

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[Serusers] Predictive Dialer
by Kapil Dhawan 28 Jul '05

28 Jul '05

Dear List Any clues on implementing some opensource Predictive Dialer with SER. Regards _________________________________________________________________ Aamir Khan is back! http://server1.msn.co.in/SP05/Mangalpandey/index.html See him in the mustached avatar in Mangal Pandey.

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sr-users July 2005