sr-users

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37456 discussions

[Serusers] checking source ip and dest ip
by İlker Aktuna (Koç.net) 20 Jun '06

20 Jun '06

Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/0B080506/07014D00/08080B4F/001…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

[Serusers] PSTN connectivity
by ravi reddy 20 Jun '06

20 Jun '06

Hi SER users , At present iam using SER-0.9.6 as proxy,registrar ,and the sip to sip calls i tested are o.k and is perfectly going and now when i tried to connect to PSTN it showing 500 internal server error i dont understand whether it is the mistake of cisco sip gateway or SER 's here is the sip messages which i get when i tried to call SIP to PSTN . I HAD GIVEN SIP TEST CLIENTS USER ID AS 32331000 AND THE LOCAL PSTN NO. ARE LIKE THIS :eg: 22458792 My Domain/Realm=81.21.33.35 and iam using onsip ser.cfg as the ser config file if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { # Domestic PSTN route(4); route(5); break; }; and the pstn connectivity lines are like this: and here are the sip messages when i tried to call pstn number my pstn gateway which iam testing now is 81.21.38.21 # U 81.21.34.34:5068 -> 81.21.33.35:5060 INVITE sip:22485400@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bKccadffff3fd60000. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>. Contact: <sip:32331001@192.168.0.74:5068>. Supported: replaces. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46532 INVITE. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 305. . v=0. o=32331001 8000 8000 IN IP4 192.168.0.74. s=SIP Call. c=IN IP4 192.168.0.74. t=0 0. m=audio 8000 RTP/AVP 8 0 18 4 2 99. a=sendrecv. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:99 iLBC/8000. a=fmtp:99 mode=20. a=ptime:20. # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bKccadffff3fd60000;received= 81.21.34.34. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=74961b5b71b6ddce908b9155b956083f.c903. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46532 INVITE. Proxy-Authenticate: Digest realm="81.21.33.35", nonce="4497c732a5a6f0ff3fb62bd9ac4561a02e4975c9". Content-Length: 0. Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27570 req_src_ip= 81.21.34.34 req_src_port=5068 in_uri=sip:22485400@81.21.33.35 out_uri= sip:22485400@81.21.33.35 via_cnt==1". . # U 81.21.34.34:5068 -> 81.21.33.35:5060 ACK sip:22485400@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bKccadffff3fd60000. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=74961b5b71b6ddce908b9155b956083f.c903. Contact: <sip:32331001@192.168.0.74:5068>. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46532 ACK. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 81.21.34.34:5068 -> 81.21.33.35:5060 INVITE sip:22485400@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>. Contact: <sip:32331001@192.168.0.74:5068>. Supported: replaces. Proxy-Authorization: Digest username="32331001", realm="81.21.33.35", algorithm=MD5, uri="sip:22485400@81.21.33.35", nonce="4497c732a5a6f0ff3fb62bd9ac4561a02e4975c9", response="af0c7b754d60bf28e493637995ac5675". Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46533 INVITE. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 305. . v=0. o=32331001 8000 8001 IN IP4 192.168.0.74. s=SIP Call. c=IN IP4 192.168.0.74. t=0 0. m=audio 8000 RTP/AVP 8 0 18 4 2 99. a=sendrecv. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:99 iLBC/8000. a=fmtp:99 mode=20. a=ptime:20. # U 81.21.34.6:5062 -> 81.21.33.35:5060 .... # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.74:5068 ;branch=z9hG4bK1be8ffffc8ecffff;rport=5068;received=81.21.34.34. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46533 INVITE. Content-Length: 0. Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27568 req_src_ip= 81.21.34.34 req_src_port=5068 in_uri=sip:22485400@81.21.33.35 out_uri= sip:22485400@81.21.38.21 via_cnt==1". . # U 81.21.33.35:5060 -> 81.21.38.21:5060 INVITE sip:22485400@81.21.38.21 SIP/2.0. Record-Route: <sip:22485400@81.21.33.35 :5060;nat=yes;ftag=26b000000d350000;lr=on>. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK7de4.ed7d9ff3.0. Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>. Contact: <sip:32331001@81.21.34.34:5068>. Supported: replaces. Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46533 INVITE. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 16. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 321. . v=0. o=32331001 8000 8001 IN IP4 192.168.0.74. s=SIP Call. c=IN IP4 81.21.33.3581.21.33.35. t=0 0. m=audio 3505035050 RTP/AVP 8 0 18 4 2 99. a=sendrecv. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:99 iLBC/8000. a=fmtp:99 mode=20. a=ptime:20. # U 81.21.38.21:5060 -> 81.21.33.35:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK7de4.ed7d9ff3.0,SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 81.21.38.21:5060 -> 81.21.33.35:5060 SIP/2.0 500 Internal Server Error. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK7de4.ed7d9ff3.0,SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 81.21.33.35:5060 -> 81.21.38.21:5060 ACK sip:22485400@81.21.38.21 SIP/2.0. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK7de4.ed7d9ff3.0. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. Call-ID: 9ad9ffff49050000(a)192.168.0.74. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. CSeq: 46533 ACK. Content-Length: 0. . # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 500 Internal Server Error. Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 81.21.34.34:5068 -> 81.21.33.35:5060 ACK sip:22485400@81.21.33.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.74:5068;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Contact: <sip:32331001@192.168.0.74:5068>. Proxy-Authorization: Digest username="32331001", realm="81.21.33.35", algorithm=MD5, uri="sip:22485400@81.21.33.35", nonce="4497c732a5a6f0ff3fb62bd9ac4561a02e4975c9", response="e09e9b23688cc3847829da66fe1fc145". Call-ID: 9ad9ffff49050000(a)192.168.0.74. CSeq: 46533 ACK. User-Agent: Grandstream HT496 1.0.2.16. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Length: 0. . # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 500 Internal Server Error. Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 81.21.33.35:5060 -> 81.21.34.34:5068 SIP/2.0 500 Internal Server Error. Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received=81.21.34.34 ;branch=z9hG4bK1be8ffffc8ecffff. From: "ravi" <sip:32331001@81.21.33.35>;tag=26b000000d350000. To: <sip:22485400@81.21.33.35>;tag=7D687D1C-202E. Date: Tue, 20 Jun 2006 09:56:36 GMT. Call-ID: 9ad9ffff49050000(a)192.168.0.74. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 46533 INVITE. Allow-Events: telephone-event. Content-Length: 0. I hope you guys will help in this issue Thanks in Advance. Regards' Ravi

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[Serusers] Compiling problem
by Alois Schafferhofer 20 Jun '06

20 Jun '06

Hi. I want to compile SER 0.9.5. Therefore, I need the packages -- libmysqlclient-dev and -- libxml2-dev The problem is, that I can't find these packages for Suse 9.3. Can someone give me an address where I can download these packages? Regards, Alois

2 1

[Users] Response to Nat problem
by Cseke Tamas 20 Jun '06

20 Jun '06

Hello! We've a problem with a provisional response. The invite is sent through NAT , and the response will be sent to incorrect port. The invite is sent from 193.77.152.140:53559 to ser, (it is a natted firewall. It not insert Via header, don't understand sip) The topmost via is: Via: SIP/2.0/UDP 10.10.1.2:5080;branch=z9hG4bK-a47f2525b20ab1a953a7c74174551819;received=193.77.152.140 The server recognize, that it is received from 193.77.152.140, but the reply will be sent to 193.77.152.140:5080 according to the RFC 3261 the port used in the response is , what present in the via header. How can i force that the response will be sent to the source port (52559)? Thanks, Tamas

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RE: [Serusers] Mediaproxy problem - how to disable P2P rtp connection
by İlker Aktuna (Koç.net) 20 Jun '06

20 Jun '06

Hi, I have found the problem. (in fact it was not a problem) The client was using the alternate media connection information which is peer-2-peer. How can I prevent P2P connection ? I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 6:13 PM To: serusers(a)iptel.org Subject: RE: [Serusers] Mediaproxy problem Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, I have found the problem. (in fact it was not a problem) The client was using the alternate media connection information which is peer-2-peer. How can I prevent P2P connection ? I need to remove "media attribute (a): alt:1" from the SIP INVITE messages, but how ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 6:13 PM To: serusers(a)iptel.org Subject: RE: [Serusers] Mediaproxy problem Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, Please help me with the following problem. Do I need SRV dns records even if I don't use load balancing ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Monday, June 19, 2006 4:31 PM To: serusers(a)iptel.org Subject: [Serusers] Mediaproxy problem Hi, I just installed mediaproxy 1.7.2 At this time, I am not interested in using more than one rtp server for loadbalancing. That's why I didn't add any DNS SRV records to my DNS. Would that be a problem ? If it's not a problem, then what's the problem described below? When I call from one client to the other one, I see that the SDP messages are modified as required (media connection infromation is changed to the media proxy server) But after first few RTP packets all RTP is routed to the alternate connection which is direct connection to the client (peer-2-peer) Thanks for your comments, ilker <http://387555.sigclick.mailinfo.com/sigclick/03090E00/07074E03/00090C49/002…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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RE: [Serusers] How to start a VOIP business ?
by Concy Pereira 20 Jun '06

20 Jun '06

Thanks Prashant, Can you give me more details about the CDR tool that comes with Asterisk. My final aim is to provide PC-to-Phone service on a small scale. Will your solution help me. Note that I am replying from a new email address Regards Concy -----Original Message----- From: Prashant Bhalesain [mailto:prashb2000@yahoo.com] Sent: Monday, June 19, 2006 11:18 PM To: Benjamin.George(a)t-systems.com; greger(a)teigre.com; CPEREIRA(a)Human-soft.com Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] How to start a VOIP business ? Boils down to Asterisk+SER I guess. You can use asterisk as a voicemail server+media server+CDR and SER as a SIP router. (connection to PSTN/termination gateway?) Asteriskathome provides most software bundled. worth giving a try. Prashant --- Benjamin.George(a)t-systems.com wrote: > Hi, > > > > I too can provide you some inputs on this apart from > the things which are already mentioned: > > > > 1. You may require a media server for playing > prompts, announcements, etc > 2. Voice mail server to store and play back voice > mails > 3. Signaling + Media Gateway if you want to connect > to PSTN world > 4. Any good performance management/monitoring tool > (normally part of OSS) to measure and manage the > performance of the entire infrastructure to assure > zero/minimum downtime > > > > Regards, > > Benjamin. > > > > > > _____ > > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Monday, June 19, 2006 1:51 PM > To: Concy Pereira > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] How to start a VOIP business > ? > > > > It's a fairly big questions you're asking. But > basically you're onto it ;-) > Ad 1. You should go for something newer than 9.0 > Ad 2. You need to decide on your maintenance > schedule and policies for upgrades/patching etc. > Many larger-scale deployments are based on SER 0.9.x > because a) it has proven extremely stable b) it is > currently maintained for bug and security fixes and > will be for a while. Others choose openser to get > more features, but you need to make sure that you > can handle testing and deployment of new versions > more often. > Ad 3. You need a RADIUS server with a user database > or you can use mysql (you need mysql regardless of > using RADIUS or not) > > g-) > > > Concy Pereira wrote: > > Sir, > > > > I would like to know how can start a successful SIP > base VOIP business. I have being going through some > documentation of Open server software but could not > come to a conclusion. > > > > Anybody can suggest me a complete solution for > running PC to Phone service Using GPL products, with > accounting, authentication and authorization with > different rate Plans etc. > > > > Here is what I have decided to do. > > > > SIP Server (Hardware) > > > > 1. Linux RedHat 9.0 > > 2. OpenSer or SIP express router > > 3. Radius client > > > > Accounting /CDR server ( Hardware) > > > > 1. Linux Redhat 9.0 > > 2. CDRTool by ag-projects > http://www.ag-projects.co > <http://www.ag-projects.co> m/cdrtool.html > > 3. MySQL > > > > 1.Carrirer > > > > Subcription to any International whole sale VOIP > carrier > > > > Please suggest if I have missed anything out. > > > > Regards, > > > > Concy > > > > > > > > _____ > > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > <http://lists.iptel.org/mailman/listinfo/serusers> > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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Re: [Users] Dumb Newbie Question
by Daniel-Constantin Mierla 19 Jun '06

19 Jun '06

On 06/19/06 20:42, Kelvin Williams wrote: > > My OpenSER box receives an Invite to > sip:+11234567890@ip.add.re.ss:5060 – my XPro is registered as > Kelvin(a)my.domain.com <mailto:Kelvin@my.domain.com>. > > When I receive this call, to my number 11234567890 – how do I make my > XPro ring? > If you have only one account, then you can do it from the script: rewriteuser("Kelvin"); lookup("location"); t_relay(); If you have many, you better use aliases. So just point 11234567890 to kelvin in aliases table (see alias_db module http://openser.org/docs/modules/1.0.x/alias_db.html) and then perform user location lookup (see registrar/usrloc modules). It will translate in: alias_db_lookup("dbaliases"); lookup("location"); t_relay(); Cheers, Daniel > kw > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] Dumb Newbie Question
by Kelvin Williams 19 Jun '06

19 Jun '06

My OpenSER box receives an Invite to sip:+11234567890@ip.add.re.ss:5060 - my XPro is registered as Kelvin(a)my.domain.com. When I receive this call, to my number 11234567890 - how do I make my XPro ring? kw

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[Serusers] publish presentity: No tuple for ....
by Alois Schafferhofer 19 Jun '06

19 Jun '06

Hi, I want my SER(0.9.3) to store status and x, y coordinates into database and to generate a NOTIFY whenever the position changes. In the log-file, I can see this information, but SER doesn't store it in the database and it doesn't generate a NOTIFY. In the log file, I get the message: publish presentity: "No tuple for ....". I hope, someone can help me. Regards, Alois /usr/sbin/ser[25182]: handle publish request /usr/sbin/ser[25182]: get_pres_uri: _puri=sip:PDA_User@fhjoanneum.at /usr/sbin/ser[25182]: handle_publish -4- p_uri= p_uri.len=23 /usr/sbin/ser[25182]: handle_publish -5- presentity=0x4069c0f8 /usr/sbin/ser[25182]: parse_pidf: sipuri=0x821b230:sip:PDA_User@fh-joanneum.at contact=(nil):(null) basic=0x821b2c8:open location=(nil):(null) /usr/sbin/ser[25182]: parse_pidf: site=(nil):(null) floor=(nil):(null) room=(nil):(null) /usr/sbin/ser[25182]: parse_pidf: x=0x821be28:01524.3739 y=0x821be38:04704.0970 radius=(nil):(null) /usr/sbin/ser[25182]: publish_presentity: no tuple for PDA_User(a)fh-joanneum.at /usr/sbin/ser[25182]: publish_presentity: event_package=1 -1- /usr/sbin/ser[25182]: handle_publish -8- paerrno=0 /usr/sbin/ser[25182]: handled publish PUBLISH sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1443248938..Max-Forwards: 70..From: <sip:PDA_User@fh-joanneum.at>;tag=271646917..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: 1040576725-1127390674-656438565..CSeq: 5 PUBLISH..Contact: <sip:PDA_User@fh-joanneum. at>..Content-Type: application/cpim-pidf+xml..Content-Length: 308..Expires:7200..Event: presence..User-Agent: SIP .NET 1.0 evaluation version, www.independentsoft.com....<?xml version="1.0" encoding="utf-8"?> <presencexmlns="urn:ietf:params:xml:ns:pidf"entity="sip:PDA_User@fh-joanneum.at"> <tupleid="9r28r59"> <status> <basic>open</basic> <geopriv><locationinfo><civilAddress><x>01524.3595</x><y>04704.0990</y></civilAddress></locationinfo></geopriv> </status> </tuple> </presence>.. SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1443248938..From: <sip:PDA_User@fh-joanneum.at>;tag=271646917..To: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-4cd6..Call-ID: 1040576725-1127390674-656438565..CSeq: 5 PUBLISH..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells:pid=25182 req_src_ip=10.0.0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1"

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[Users] nat traversal
by Padmaja RV 19 Jun '06

19 Jun '06

Hi all, i need to implement openser for nat traversal. can anyone tell me if using nat helper module would be better or using media proxy. from so far i read, i understood that mediaproxy needs separate installation but does not need to make much change in the existing openser.cfg. where as if i use the nathelper, there is no need for separate media proxy installation but the openser cfg needs to be changed. Please suggest the best way to do it....? also please correct me if what i understood is not right...... Thanks so much, Padmaja

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