sr-users

sr-users@lists.kamailio.org

42 participants
37519 discussions

[Users] Re: [Devel] New module for OpenSER - Application Server interconnection
by Bogdan-Andrei Iancu 22 Dec '06

22 Dec '06

Hi Elias, (also fwd. to "users" to let everybody know) thanks for the contribution. I think that the new SEAS and PERL modules will give a lot more options when comes to integrate OpenSER with other application and what is more important , they will allow non-C extensions :) !! I will generate the docs and place them on the web also. BTW, are some default application that I can give a fast try? I would like to test some real-case apps to get the feeling. Regards and Merry Christmas, Bogdan Elias Baixas wrote: > Hello everybody, > > I just submitted a new module, called SEAS (which stands for Sip > Express Application Server). > This module allows a third party to create an Application Server and > connect it to OpenSER to interoperate seamlessly. > This permits the Application Server to trigger actions on OpenSER, > such as sending out a request, or replying a UAS transaction, in some > way similarly to what can be achieved using fifo, or MI commands > "t_uac_dlg" and "t_reply", but SEAS provides a much richer API to this > kind of tasks. > The module also allows OpenSER to notify the Application Server when > requests/responses come in, and then transfer the control over the > incoming request entirely to the Application Server. > Furthermore, the module implements a binary protocol that transfers > the SIP message along with information abouth the structure of the > message's headers (ie. where each header starts, and how long it is), > so the Application Server doesn't need to reparse the entire SIP > message, thus improving Application Server's performance very much. > The SEAS protocol also carries information about the transaction being > processed at OpenSER for every message, so that the Application Server > doesn't need to implement all the SIP-transaction machinery, thus > actually profitting from the highly optimized TM module at OpenSER. > > The Application Server can be programmed in any language, because the > SEAS module comunicates with it through a TCP socket. > > At the moment, SEAS module has been written to work with an > Application Server developed by VozTelecom. It is called WeSIP and can > be downloaded freely from http://www.wesip.eu . WeSIP is an > implementation of the SipServlet and HttpServlet standards, so it > allows a programmer to easily implement Web-Sip converged applications > (such as Click2call). > WeSIP is programmed entirely in Java, so you have all the tools and > facilities from Java and J2EE available to program SIP applications, > while already using OpenSER to process the SIP messages and > transactions at the low level, thus profitting both from the high > performance of OpenSER, and the high-level SIP-Application programming > framework which is SipServlet and the JAVA languaje. > > I hope this module will help create new usage scenarios for OpenSER, > allowing it to provide high-level Telephony Applications to users. > > The module is still in a alpha stage, but should be able for > production shortly. > > > best regards, > > Elias Baixas > > VozTelecom, Sistemas > http://www.wesip.eu > http://www.voztele.com > > _______________________________________________ > Devel mailing list > Devel(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/devel >

3 3

[Serusers] Invite going to my gateway
by Bill 22 Dec '06

22 Dec '06

Scenario: I have two test phones setup to a SER server. The two phones are 192.168.2.133 and 192.168.2.101. My server is 192.168.2.1. I can call from .101 to .133 fine. When I try to place a call from .133 this is what I get. 0(2734) SIP Request: 0(2734) method: <INVITE> 0(2734) uri: <sip:1001@192.168.2.1> 0(2734) version: <SIP/2.0> 0(2734) parse_headers: flags=1 0(2734) Found param type 232, <branch> = <z9hG4bK7e2902f2a>; state=16 0(2734) end of header reached, state=5 0(2734) parse_headers: Via found, flags=1 0(2734) parse_headers: this is the first via 0(2734) After parse_msg... 0(2734) preparing to run routing scripts... 0(2734) parse_headers: flags=128 0(2734) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2734) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2734) parse_headers: flags=8 0(2734) DEBUG: get_hdr_body : content_length=586 0(2734) end of header reached, state=9 0(2734) DEBUG: get_hdr_field: <To> [29]; uri=[sip:1001@192.168.2.1] 0(2734) DEBUG: to body [1001 <sip:1001@192.168.2.1> ] 0(2734) DEBUG: add_param: tag=4d4e374f5ebe6ab 0(2734) end of header reached, state=29 0(2734) parse_headers: flags=256 0(2734) get_hdr_field: cseq <CSeq>: <1338505295> <INVITE> 0(2734) is_preloaded: Yes 0(2734) grep_sock_info - checking if host==us: 13==13 && [192.168.2.136] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) after_loose: Topmost route URI: 'sip:192.168.2.136:5060;lr' is me 0(2734) parse_headers: flags=256 0(2734) found end of header 0(2734) find_next_route: No next Route HF found 0(2734) after_loose: No next URI found 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) check_self: host != me 0(2734) DEBUG: t_newtran: msg id=45 , global msg id=43 , T on entrance=0xffffffff 0(2734) parse_headers: flags=-1 0(2734) parse_headers: flags=60 0(2734) t_lookup_request: start searching: hash=15747, isACK=0 0(2734) DEBUG: RFC3261 transaction matching failed 0(2734) DEBUG: t_lookup_request: no transaction found 0(2734) SER: new INVITE 0(2734) parse_headers: flags=-1 0(2734) check_via_address(192.168.2.133, 192.168.2.133, 0) 0(2734) WARNING:vqm_resize: resize(0) called 0(2734) DEBUG: reply sent out. buf=0x8104e98: SIP/2.0 1..., shmem=0xb6145728: SIP/2.0 1 0(2734) DEBUG: _reply_light: finished 0(2734) DEBUG: mk_proxy: doing DNS lookup... 0(2734) check_via_address(192.168.2.133, 192.168.2.133, 0) 0(2734) DEBUG: add_to_tail_of_timer[4]: 0xb6143eb8 0(2734) DEBUG: add_to_tail_of_timer[0]: 0xb6143ec8 0(2734) SER: new transaction fwd'ed 0(2734) DEBUG:destroy_avp_list: destroying list (nil) 0(2734) receive_msg: cleaning up 1(2737) DEBUG: timer routine:4,tl=0xb6143eb8 next=(nil) 1(2737) DEBUG: retransmission_handler : request resending (t=0xb6143da0, INVITE si ... ) 1(2737) DEBUG: add_to_tail_of_timer[5]: 0xb6143eb8 1(2737) DEBUG: retransmission_handler : done 0(2734) SIP Request: 0(2734) method: <CANCEL> 0(2734) uri: <sip:1001@192.168.2.1> 0(2734) version: <SIP/2.0> 0(2734) parse_headers: flags=1 0(2734) Found param type 232, <branch> = <z9hG4bK7e2902f2a>; state=16 0(2734) end of header reached, state=5 0(2734) parse_headers: Via found, flags=1 0(2734) parse_headers: this is the first via 0(2734) After parse_msg... 0(2734) preparing to run routing scripts... 0(2734) parse_headers: flags=128 0(2734) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2734) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2734) parse_headers: flags=8 0(2734) DEBUG: get_hdr_body : content_length=0 0(2734) end of header reached, state=9 0(2734) DEBUG: get_hdr_field: <To> [29]; uri=[sip:1001@192.168.2.1] 0(2734) DEBUG: to body [1001 <sip:1001@192.168.2.1> ] 0(2734) DEBUG: add_param: tag=4d4e374f5ebe6ab 0(2734) end of header reached, state=29 0(2734) parse_headers: flags=256 0(2734) get_hdr_field: cseq <CSeq>: <1338505295> <CANCEL> 0(2734) is_preloaded: Yes 0(2734) grep_sock_info - checking if host==us: 13==13 && [192.168.2.136] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) after_loose: Topmost route URI: 'sip:192.168.2.136:5060;lr' is me 0(2734) parse_headers: flags=256 0(2734) found end of header 0(2734) find_next_route: No next Route HF found 0(2734) after_loose: No next URI found 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) check_self: host != me 0(2734) DEBUG: t_newtran: msg id=46 , global msg id=45 , T on entrance=0xffffffff Bill Hissong

4 3

[Users] re: how to pass the Caller ID into uac_replace_from function ?
by don lin 22 Dec '06

22 Dec '06

Hi there, I saw all of you discussed about the postgres and I do have a question which somewhat related to postgres as well (please refer to the end of this email for detail). My question is that I have Caller ID (regular phone number) mapping to a SIP account stored in postgres, when I call from SIP device to a PSTN phone, telco carrier does not accept the SIP From field and requested to use regular phone number. I use exec_msg to call the external C program which read the Caller ID mapping to SIP account. However, I do not know how to pass this Caller ID into uac_replace_from. Were the postgres version work for this purpose ? Please advise if you have any suggestion. Thanks in advance, Don ------------------------------ Message: 5 Date: Thu, 21 Dec 2006 12:20:47 +0200 From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> Subject: Re: [Users] postgres To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: users(a)openser.org Message-ID: <458A5FFF.4060003(a)voice-system.ro> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello Klaus, On 12/21/06 12:05, Klaus Darilion wrote: > Hi Daniel! > > I've checked the postgres script: rel_1_1_0 works fine, but there is a > problem in unstable: The domain for the default admin user is asked > not for serweb tables but for the plain openser tables as well. Also > the subscriber table includes all the serweb columns even if serweb > tables are not installed. I think a merge with mysql version, with an option to say the underlaying DB engine is the right solution, to be easy to maintain -- same as we did with openserctl. I will have that in mind, but the best will be to register it to tracker. Cheers, Daniel > > regards > klaus > > > Daniel-Constantin Mierla wrote: >> Hello, >> >> On 12/20/06 19:36, Mark Price wrote: >>> >>> >>> On 12/20/06, *Klaus Darilion* < klaus.mailinglists(a)pernau.at >>> <mailto:klaus.mailinglists@pernau.at>> wrote: >>> >>> Hi Mark! >>> >>> Postgres should work well - I use it since ser 0.8. Just make sure >>> that >>> the hard disk does not get full, because this breaks the index >>> inside >>> postgres and postgres is getting real slow (re-create the index >>> if it >>> happens). >>> >>> Last time I tested openser_postgres.sh it worked fine. If you find >>> a bug >>> please let us know. >>> >>> Please check to use the latest versions from CVS (for 1.1 use CVS >>> rel_1_1_0) >>> >>> >>> The latest version from cvs doesn't work out of the box with openser >>> build from the release tarball, because the release tarball doesn't >>> included /usr/sbin/openser_gen_ha1 (although I could get past this >>> by replacing it with the md5sum equivelant). >>> >>> However, the release tarball doesn't work because of this bug: >>> https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_… >>> <https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_…> >>> >>> >>> So, 1. is the cvs version of rel_1_1_0 considered safe for >>> production use? >> >> yes, it is the most recommended version to use -- several issues were >> found and fixed since 1.1.0 release. >> >>> 2. if so and there are important fixes such as this in CVS, >>> would openser.org <http://openser.org> consider >>> releasing a point release containing such changes? >> It is an option, and perhaps should be taken in consideration, as >> patch release with not so much packaging. >> >> This may help some which want to maintain platforms updated, it is >> exampled with 1.0.x, bust just replace the version with 1.1.0 in the >> document and should work (rel_1_0_0 => rel_1_1_0) >> http://openser.org/dokuwiki/doku.php/install:openser-from-cvs >> >> Cheers, >> Daniel >> >> >>> Thanks, >>> >>> Mark Price don lin <don12lin(a)yahoo.com> wrote: Hi there, We have a Postgres Database to store the users' informations before we use Openser. To integrate between Postgres and Openser, we write external C programs and call them from Openser with exec_dset function. The issue we have now is with the Caller ID. When the user makes a call to PSTN number from SIP devices (soft phone or hard phone), the telco carrier requests us to send a numerical Caller ID, not the SIP format (such as john.dow(a)abc.com). What I am doing is to add the Caller ID field into the Postgres for each user and write a C program to query the Caller ID field, then Openser calls it with exec_dset functioin. However, I did not find a way to define a variable in Openser to store this Caller ID and pass it to uac_replace_from function. Appreciate for any suggestions. Jeff, You mentioned that you replaced the From Header, may I ask how you do it ? Thanks and Regards, Don ------------------------------ <Message: 4 <Date: Tue, 19 Dec 2006 11:11:09 +0900 <From: Jeff Williams <jeffw(a)globaldial.com> <Subject: [Users] (PR)ACK problem with uac_replace_from <To: users(a)openser.org <Message-ID: <45874A3D.5070206(a)globaldial.com> <Content-Type: text/plain; charset="iso-8859-1" <I seem to have an issue with uac_replace_from. <I am using openser as a media proxy. For invites I replace the numbers <with leading 0's with 61 for Australia and forward the call to our LCR <box which then forwards the call to the correct voice to pstn gateway. <I am trying to use uac_replace_from to set the from address so the <callerid appears correct on the outgoing calls. <This seems to work fine for the INVITE request, but on subsequent PRACKs <and ACKs (which just get loose routed), something strange happens to <the from address. <On invite, the From get re-written: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <sip:0892209080@sipone.com>;tag=1790577941 <For the PRACKs and ACKs, this happens: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <<\032\317\352S\232\222B\230\234K\231*BV\300\030\303\332\ah\303\306i\365\005\016\200\250\000\000\230\016\200\252j\200\002`\000\000\016>;tag=1790577 <941 <There shouldn't be anything happening to the From address on the <(PR)ACKs at all. <Anyone have any ideas? <I have attached a ethereal dump of the call setup. <Jeff __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 2

[Users] uri, myself, etc., variables, + is_domain_local()
by Mark Price 22 Dec '06

22 Dec '06

Hi, I've got a few questions: 1. where can I find out what all the variables like uri, myself, etc., mean? 2. how come is_domain_local($ruri) has a different result than uri==myself? In particular, I have my external IP address defined as an alias. I.e.: alias=4.2.2.2 When I say if( ! is_domain_local($ruri) ) when $ruri = user(a)4.2.2.2, the check succeeds. When I say if( ! uri==myself ), the check fails (correctly, I think). Thanks, Mark Price

2 1

[Serusers] mysql tables help
by Katty Xiong 22 Dec '06

22 Dec '06

I am tring to use Ottendorf. I have db tables for 0.9.6 on the machine, so I decide to drop the old tables and create new tables using ser_mysql.sh for Ottendorf, but the script keeps on giving me errors. Do I have to delete the entire old version of SER (0.9.6)? thanks, Joy __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] help needed on avpops
by Zoran Milic 22 Dec '06

22 Dec '06

Hi guys, I am trying to set up radius and avp for ages now, and this time i think i am close to it. I have a problem and I can't find any information about it. The thing is, my soft agent (set to uid 101) can't register, and it is getting "Forbidden IP" even with this in the cfg file (see below) I have set up freeradius and radiusclient-ng according to the manual on http://www.openser.org/docs/openser-radius-1.0.x.html. I can't find anything on the error i am getting in my log. If you need my complete ..cfg or log file, just ask, and I will send them promptly. thanks in advance, best regards, Zoran openser.cfg: .... xlog("L_ERR","rpid = $avp(s:rpid)"); xlog("L_ERR","adresa = $avp(s:adresa)"); xlog("L_ERR","adresa2 = $avp(s:adresa2)"); # check the src ip address # avp_load_radius("caller"); if(!avp_check("$adresa", "eq/$adresa/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; .... freeradius/users: .... 101 Auth-Type := Digest, User-Password == "101" Reply-Message = "Authenticated", Sip-Avp += "rpid:101", Sip-Avp += "adresa:10.0.0.157", Sip-Avp += "adresa2:10.0.0.157" .... last entries in the log after the rejected register. .... DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa2:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa2> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa2'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <rpid:101> DEBUG:auth_radius:extract_avp: AVP name is <rpid> DEBUG:auth_radius:extract_avp: AVP val is <101> DEBUG:auth_radius:generate_avps: AVP 'rpid'/0='101'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa2:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa2> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa2'/0='10.0.0.157'/0 has been added rpid = 101 adresa = 10.0.0.157 adresa2 = 10.0.0.157 xl_get_spec_value: error - null sp->itf ERROR:avpops:ops_check_avp: cannot get src value parse_headers: flags=ffffffffffffffff check_via_address(192.168.52.145, 192.168.52.145, 0) DEBUG:destroy_avp_list: destroying list 0x404d9238 receive_msg: cleaning up

2 2

Re: [Users] OpenSER as load balancer for several Asterisk servers
by Edoardo Serra 22 Dec '06

22 Dec '06

Hi Users, I'm here again. After having read more docs I don't feel like using rtpproxy My Asterisks have both public IP Adresses so I don't need to proxy rtp stream I just need a configuration to randomly proxy SIP requests to Asterisk boxes behind SER (for load balancing purposes) and to handle registrations of user to forward incoming calls to them (calls are coming from Asterisks to OpenSER) I'm really stuck on it because I already have such a working setup, with same config files. I'm just trying to replicate it on another site Tnx in advance for help Edoardo My last mail to the ml follow for reference Tnx again Klaus, i'll read docs more in depth and try to understand each change to openser.cfg you suggested >>U 2006/12/20 19:15:35.025446 OOO.OOO.OOO.OOO:5060 -> CCC.CCC.CCC.CCC:21722 >>SIP/2.0 183 Session Progress. >>Via: SIP/2.0/UDP >>OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-3e229802603d7c32-1--d87543-;rport=21722. >> > >^^^^^^^^^^^^^^ > >strange bug again - there must be CCC.CCC.CCC.CCC really strange, do you think it's an openser bug ? The strangest thing is that I just copied openser configs from a working system (Openser + Asterisks) changing just ip addresses OpenSER version is also the same... [...] >># !! Nathelper >>onreply_route[1] { >> # NATed transaction ? >> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { >> fix_nated_contact(); >> #force_rtp_proxy(); > ^ >Are you sure it is commented? I do not believe it because the ngrep >shows that the SDP of 200 Ok is rewritten. Sure, I also tried to remove the line without solving the previous problem >Please read the Getting Started Turial from onsip.org carefully. It >describes how you handle NAT correctly and also describes how to do >NAT traversal for in-dialog messages, which is missing in your config. I will. Tnx again Regards Edoardo

1 1

Re: [Users] Load-Balancing
by raviprakash sunkara 22 Dec '06

22 Dec '06

May i guess U Perform the Dispatcher modules ..... May this Hopes u to get On 12/22/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > > > > > On 12/22/06, Thomas Deillon < Thomas.Deillon(a)smart-telecom.ch> wrote: > > > > Hi, > > > > I'm trying to set a load balancing solution with SER and twos Asterisk. > > The Asterisks will be use as REGISTRAR, REDIRECT, GATEWAY .... > > > > The ser will just load balance the traffic on one Asterisk or an other. > > I use the dispatcher module for the SIP message and I use 2 public ip > > addresses for the asterisk and so, I'm able to directly send the RTP > > traffic to them (I didn't use mediaproxy or rtpproxy). > > > > On each asterisk, I set the SER as default GW to be sure that all > > message go through this computer. (Client are behind a FW/NAT and so, I > > would like to be sure that all message come from the same IP > > address:port else, the FW/NAT will DROP these messages). > > > > For the moment, with this solution, I have a problem. The message from > > client will go to SER which will resend it to the asterisk and then, the > > > > asterisk will try to answer responding on the same IP -> Loop detected > > !! > > > > On some device I saw that we can forward the request for the phone to > > the asterisk without changing the source ip address .... > > > > Do you know how can I do this ? > > > > Thanks a lot for your help, > > > > Thomas D > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com > www.hyperion-tech.com -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

1 0

[Users] Routing calls to a sip terminator
by Xantek, Inc. 22 Dec '06

22 Dec '06

I am sure this must have been covered somewhere, but I cannot find it. I am trying to get openser to register with username and password and forward a call to that server. This is for outbound sip termination to the PSTN. If somebody could point me to the right place, I would be grateful. Bill

7 8

[Users] Load-Balancing
by Thomas Deillon 21 Dec '06

21 Dec '06

Hi, I'm trying to set a load balancing solution with SER and twos Asterisk. The Asterisks will be use as REGISTRAR, REDIRECT, GATEWAY .... The ser will just load balance the traffic on one Asterisk or an other. I use the dispatcher module for the SIP message and I use 2 public ip addresses for the asterisk and so, I'm able to directly send the RTP traffic to them (I didn't use mediaproxy or rtpproxy). On each asterisk, I set the SER as default GW to be sure that all message go through this computer. (Client are behind a FW/NAT and so, I would like to be sure that all message come from the same IP address:port else, the FW/NAT will DROP these messages). For the moment, with this solution, I have a problem. The message from client will go to SER which will resend it to the asterisk and then, the asterisk will try to answer responding on the same IP -> Loop detected !! On some device I saw that we can forward the request for the phone to the asterisk without changing the source ip address .... Do you know how can I do this ? Thanks a lot for your help, Thomas D

1 0

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