sr-users

sr-users@lists.kamailio.org

10 participants
37525 discussions

[Serusers] error while b2b installation
by rupesh boyana 30 Nov '06

30 Nov '06

hi, after downloading latest b2b from vovida (sip-1.5.0.tar.gz), when i tried to install it, got the following error : In file included from Timer.cxx:58: Timer.h: In member function `bool Timer<T>::sleepFor(timeval*)': Timer.h:230: error: `cout' was not declared in this scope Timer.h:230: error: `endl' was not declared in this scope Timer.h: In member function `void Timer<T>::insert(T, int)': Timer.h:267: error: `cout' was not declared in this scope Timer.h:267: error: `endl' was not declared in this scope make[1]: *** [obj.debug.Linux.i686/Timer.o] Error 1 make[1]: Leaving directory `/root/sip-1.5.0/util' make: *** [util] Error 2 Initially i gave ./configure inside the sip-1.5.0 folder. Then when i tried with make b2bua , got the following error and installation stopped. i have ser with mysql,nathelper,rtp proxy. thank you, Rupesh.

2 1

[Users] auth_db subscriber caching
by Klaus Darilion 30 Nov '06

30 Nov '06

Hi! A stupid question: Is the subscriber table also cached inside openser (like location table)? regards klaus -- Klaus Darilion nic.at

2 5

[Serusers] SER/Asterisk inbound problem [corrected]
by Sameer Ahmed 30 Nov '06

30 Nov '06

The Setup: I have two asterisk servers, AsterA and AsterB, and one ser server acting as a proxy. Both AsterA and AsterB are connected to Ser as sip clients. On the asterisk servers I have defined a SIP trunk. The idea is that users/extensions on AsterA should be able to reach users/extensions on AsterB via Ser. The Problem: My extension numbers on AsterA range from 201 to 220. On AsterB I also have the same range of extensions, 201 to 220. When user 205 on AsterA places a call to user 201 on AsterB, he gets a network congestion message. Looking at the sip packets in Asterisk CLI, I get a "Forbidden" message from AsterB. Here's the interesting bit: When I create a user/extension on AsterA that does not exist on AsterB, like 401, I do not get this problem and the call goes through just fine. Apparently, the problem has to do with the fact that I am using the same list of extensions on both asterisk servers. Detailed Configuration: All three servers are on the same network. IP Address of AsterA: 192.168.0.9 IP Address of AsterB: 192.168.0.14 IP Address of SER: 192.168.0.15 To dial out from either asterisk server using the SIP trunk: 91 must be entered before the number SER must know which server the call is meant for, so the following codes are used by SER to distinguish between calls for AsterA and AsterB: AsterA: 52 AsterB: 51 Example 1: If 205 on AsterA wants to call 201 on AsterB, he must dial 9151201 Example 2: If 201 on AsterB wants to call 205 on AsterA, he must dial 9152205 I have insecure=invite in both sip.conf files, for AsterA and AsterB Below are the sip packets I picked up at AsterA (192.168.0.9) when 9152201 was dialed by extension 205 at AsterB (192.168.0.14) <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15 ;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: < sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9000 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1371cb5b" Content-Length: 0 --- Scheduling destruction of call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14' in 15000 ms Found user '205' <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: < sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 102 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15 ;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: < sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Proxy-Authorization: Digest username="bf51", realm="asterisk", algorithm=MD5, uri="sip:52201@192.168.0.15", nonce="1371cb5b", response="8ecd1a24c31a02fc938308a0f5c48924", opaque="" Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9001 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (16 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Found user '205' Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" < sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15 ;branch=z9hG4bKcb71.18308a81.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 103 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14'

1 0

[Serusers] SER/Asterisk inbound problem
by Sameer Ahmed 30 Nov '06

30 Nov '06

The Setup: I have two asterisk servers, AsterA and AsterB, and one ser server acting as a proxy. Both AsterA and AsterB are connected to Ser as sip clients. On the asterisk servers I have defined a SIP trunk. The idea is that users/extensions on AsterA should be able to reach users/extensions on AsterB via Ser. The Problem: My extension numbers on AsterA range from 201 to 220. On AsterB I also have the same range of extensions, 201 to 220. When user 205 on AsterA places a call to user 201 on AsterB, he gets a network congestion message. Looking at the sip packets in Asterisk CLI, I get a "Forbidden" message from AsterB. Here's the interesting bit: When I create a user/extension on AsterA that does not exist on AsterB, like 401, I do not get this problem and the call goes through just fine. Apparently, the problem has to do with the fact that I am using the same list of extensions on both asterisk servers. Detailed Configuration: All three servers are on the same network. IP Address of AsterA: 192.168.0.9 IP Address of AsterB: 192.168.0.14 IP Address of SER: 192.168.0.15 To dial out from either asterisk server using the SIP trunk: 91 must be entered before the number SER must know which server the call is meant for, so the following codes are used by SER to distinguish between calls for AsterA and AsterB: AsterA: 51 AsterB: 52 Example 1: If 205 on AsterA wants to call 201 on AsterB, he must dial 9152201 Example 2: If 201 on AsterB wants to call 205 on AsterA, he must dial 9151205 I have insecrure=invite in both sip.conf files, for AsterA and AsterB Below are the sip packets I picked up at AsterA (192.168.0.9) when 9152201 was dialed by extension 205 at AsterB (192.168.0.14) <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: <sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9000 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK3170503d;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1371cb5b" Content-Length: 0 --- Scheduling destruction of call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14' in 15000 ms Found user '205' <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKbb71.5d6dc223.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 102 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.0.15:5060: INVITE sip:201@192.168.0.9 SIP/2.0 Record-Route: <sip:192.168.0.15;lr=on;ftag=as0ebc5aac> Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15> Contact: <sip:205@192.168.0.14> Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Proxy-Authorization: Digest username="bf51", realm="asterisk", algorithm=MD5, uri="sip:52201@192.168.0.15", nonce="1371cb5b", response="8ecd1a24c31a02fc938308a0f5c48924", opaque="" Date: Thu, 30 Nov 2006 11:25:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 9000 9001 IN IP4 192.168.0.14 s=session c=IN IP4 192.168.0.14 t=0 0 m=audio 19704 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (16 headers 11 lines) --- Using INVITE request as basis request - 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 Sending to 192.168.0.15 : 5060 (non-NAT) Found user '205' Reliably Transmitting (no NAT) to 192.168.0.15:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0;received= 192.168.0.15 Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK73fc192a;rport=5060 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 192.168.0.15:5060: ACK sip:201@192.168.0.9 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.15;branch=z9hG4bKcb71.18308a81.0 From: "205" <sip:205@192.168.0.14>;tag=as0ebc5aac Call-ID: 3f282c4a48f8ec96669305565da7980a(a)192.168.0.14 To: <sip:52201@192.168.0.15>;tag=as2a7b6cfc CSeq: 103 ACK User-Agent: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '3f282c4a48f8ec96669305565da7980a(a)192.168.0.14'

1 0

RE: [Users] Multiple register message
by Thomas Deillon 30 Nov '06

30 Nov '06

Thanks a lot. It's work. Thomas -----Message d'origine----- De : Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Envoyé : mercredi, 29. novembre 2006 19:24 À : Thomas Deillon Cc : Klaus Darilion; users(a)openser.org Objet : Re: [Users] Multiple register message Hi Thomas, your script will not correctly work as it is a combination of stateless and stateful processing. t_replicate, t_on_failure() are stateful functions, while forward() does stateless fwd - try t_relay() from TM module. regards, bogdan Thomas Deillon wrote: >Subject: Trying to find a solution to duplicate an incoming REGISTER to >two asterisks boxes. > >1. maybe Asterisk can use the location table of openser too? >2. you can save the contact with save_no_reply and then t_relay to the >asterisk server, or > >1&2 > In fact, SER didn't register phones. It just make forwards > >3. you can use t_replicate ... > > >>I tried but I didn't find the way to do this. >> >> > >I have the configuration below : > > >route{ > > if(method=="REGISTER"){ > t_replicate("192.168.1.119") > forward("192.168.1.120") > } > > ds_select_dst("0", "4"); > forward(); > t_on_failure("1"); >} > >failure_route[1] { > log("next_dst\n"); > ds_next_dst(); >} > > >Thanks a lot for your help. >Thomas > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

[Serusers] how to get source uri?
by jorain 30 Nov '06

30 Nov '06

Hi all, How to get the source uri from exec function? like SIP_USER, SIP_RURI and so on. SIP_HF_FROM give full path, including source username, uri and tag. I just want to get the source uri. Please guide. Thanks for any help. Regards, jorain

1 0

Re: [Users] modifying From/To headers with uac module?
by Mahesh Paolini-Subramanya 30 Nov '06

30 Nov '06

Good to know! Strangely enuff, I got the same response from a handful of people too :-) FWIW, I'm probably still going to mess around with getting uac_replace_to operational, 'cos it does seem to be somewhat useful (especially with some of the stranger carrier requirements that I'm starting to see out there) cheers ----- Original MessagGe ----- From: T.R. Missner <tmissner(a)gmail.com> To: mahesh(a)aptela.com Cc: users openser.org <users(a)openser.org> Sent: Wednesday, November 29, 2006 3:35:47 PM GMT-0600 Subject: Re: [Users] modifying From/To headers with uac module? as a former level3 engineer who wrote some of level3s voip platform parts and current customer i can tell you with certainty that level3 only enforces the e.164 requirement on the from and to header during interop. if you can mock up your headers long enough to pass interop and production turn up you can then safely terminate traffic to l3 without e.164 formating. they will not refuse your calls once you make it to production. of course a better solution may be choosing a meta carrier like bandwidth.com ( where i work ) hope this helps tr On 11/29/06, Mahesh Paolini-Subramanya <mahesh(a)corp.aptela.com> wrote: > FWIW, i've been bemoaning the lack of 'To' rewriting for a while myself. > Level3, in its infinite wisdom, has decided that they need the RURI, From, > and 'To' headers to be in a very specific format (prepended '+', 10 digits, > etc., etc.) > I can get the RURI and 'From' exactly the way they need it, but the 'To' is > definitely beyond my ability. > > Do let me/us know if you intend to go ahead with 'uac_replace_to'. It would > be trememdously useful... > > cheers > ----- Original Message ----- > From: Alan Crosswell <alan(a)columbia.edu> > To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Cc: users openser.org <users(a)openser.org> > Sent: Tuesday, November 28, 2006 3:35:22 PM GMT-0600 > Subject: Re: [Users] modifying From/To headers with uac module? > > Thanks Bogdan. I'll have to double-check with them; They may have only > meant they wanted the From in E.164 so that Caller ID works correctly. > /a > > Bogdan-Andrei Iancu wrote: > > Hi Alan, > > > > my advice is to change the PSTN termination since their service is not > > RFC compliant - TO header has no use in routing, only the RURI being > > used (according to the RFC 3261 ~ 3 years old). > > > > but to answer to your question, yes that is the proper place and the > > mechanism is 95% the same as for FROM hdr. > > > > regards, > > bogdan > > > > Alan Crosswell wrote: > > > >> Hello, > >> > >> I see that UAC allows modifying the From header, but I would also like > >> to modify the To. This is because an ITSP I will be testing with wants > >> the From and To to be written in E.164 form (rewriting the R-URI appears > >> not to be good enough) while my internal registrations use a 5-digit > >> extension. It looks like uac is the right module for this but it > >> appears that it can only modify the From header with uac_replace_from() > >> and uac_restore_from(). Would this be the right place to add > >> uac_replace_to() and uac_resotre_to() as well? > >> > >> /a > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)openser.org > >> http://openser.org/cgi-bin/mailman/listinfo/users > >> > >> > >> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > > -- > ******************************************* > Mahesh Paolini-Subramanya (703) 386-1500 x9100 > CTO mahesh(a)aptela.com > Aptela, Inc. http://www.aptela.com > "Aptela: How Business Answers The Call" > ******************************************* > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

2 1

[Users] modifying From/To headers with uac module?
by Alan Crosswell 29 Nov '06

29 Nov '06

Hello, I see that UAC allows modifying the From header, but I would also like to modify the To. This is because an ITSP I will be testing with wants the From and To to be written in E.164 form (rewriting the R-URI appears not to be good enough) while my internal registrations use a 5-digit extension. It looks like uac is the right module for this but it appears that it can only modify the From header with uac_replace_from() and uac_restore_from(). Would this be the right place to add uac_replace_to() and uac_resotre_to() as well? /a

4 4

RE: [Users] Multiple register message
by Thomas Deillon 29 Nov '06

29 Nov '06

Subject: Trying to find a solution to duplicate an incoming REGISTER to two asterisks boxes. 1. maybe Asterisk can use the location table of openser too? 2. you can save the contact with save_no_reply and then t_relay to the asterisk server, or 1&2 > In fact, SER didn't register phones. It just make forwards 3. you can use t_replicate ... > I tried but I didn't find the way to do this. I have the configuration below : route{ if(method=="REGISTER"){ t_replicate("192.168.1.119") forward("192.168.1.120") } ds_select_dst("0", "4"); forward(); t_on_failure("1"); } failure_route[1] { log("next_dst\n"); ds_next_dst(); } Thanks a lot for your help. Thomas

2 1

[Users] Re: Billing system with good integration with OpenSER
by M.Emran 29 Nov '06

29 Nov '06

u can check iSoftSwitch.pls visit www.inspiresoftbd.com -- Regards ---------- M Emran InSpiration Software Ltd. E-mail:monemran@gmail.com IM: mohammademran(a)hotmail.com Web: www.inspiresoftbd.com

2 1

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