Re: [OpenSIPS-Users] Opensips as SBC
Thank you for response, Actually what I'm looking for is relaying and controlling the traffic according to static/dynamic rules that take into consideration what's available in the SIP header or by taking into account link conditions or status if possible. So, it's a kind of RTP relaying through different links/interfaces and according to different rules. Regards. On Wed, 2009-09-30 at 22:47 +0300, Bogdan-Andrei Iancu wrote: Hi Ghaith, an SBC is a very generic term.it can do a lot of stuff (NAT , topi hiding, net bridging, security, etc)... So, what kind of functionalities you have in mind when you ask about a SBC? Regards, Bogdan Ghaith ALKAYYEM wrote: Hello, Do you think that we can consider OpenSIPS as a real SBC? and if not what do you think the missed functionalities are? or one has to add some module in Opensips to act it as SBC are there some performance drawbacks and/or other issues while converting Opensips to SBC? thanks for your reply in advance. Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] static
Hi, Could you tell me why the most functions and structures in the developed modules are static? Regards. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips as SBC
Hello, Do you think that we can consider OpenSIPS as a real SBC? and if not what do you think the missed functionalities are? or one has to add some module in Opensips to act it as SBC are there some performance drawbacks and/or other issues while converting Opensips to SBC? thanks for your reply in advance. Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Diameter
Thank you for response, I see in the details of that module (auth_diameter) this diagram: ++ SIP INVITE +=+ DIAMETER +--+ +--+ || no Auth hdr #/# AA-Request| | | | ||-1---#/#---2---| |---2--| | |UAC | #UAS//#|DClnt | |DSrv | ||-4---#(SER)#--3|(DISC)|--3---|(DISC)| || 401 #/# DIAMETER | | | | ++ Unauthorized +=+ AA-Answer +--+ +--+ We notice in this architecture that we have two diameter blocks, the first one plays the role of diameter client(DClnt) and the second one plays the role of diameter server(DSrv). But in Radius modules the OpenSIPS interacts with Radius server directly, so maybe I have a misunderstood in this regard but I'd like to know whether it's possible to make OpenSIPS interact with Diameter server directly or this is not possible due to the nature of diameter protocol. Opendiameter is written in C++ so I think it's not possible to integrate it directly in OpenSIPS as a module, so we have to design something similar to the above diagram, isn't it? What would be the type of communication between OpenSIPS Diameter Client, is it diameter based also? The implementation of Openblox looks promising as well, so do you think it would be a good candidate for building the module? Regards. On Tue, 2009-09-22 at 14:17 +0300, Bogdan-Andrei Iancu wrote: Hi Ghaith, Ghaith ALKAYYEM wrote: Hello lists, I'm interested in AAA functions according to Diameter which is newer than Radius. yes, the new AAA interface will simplify a lot the addition of DIAMETER in OpenSIPS. All modules using the AAA interface will be automatically able to use the DIAMETER support. There's a module in OpenSIPS which is called AUTH_DIAMETER Module and it's mentioned that this module is obsolete. yes ,it is obsolete as it is using an old and obsolete DIAMETER client-server implementation (DISC). So I'd like your recommendations about this matter, should I work from the scratch to develop something that does this functionalities or is it possible to integrate other open source software with OpenSIPS. Our plan is to use some opensource libraries to build a DIAMETER (aaa_diameter module) implementation for the AAA API in OpenSIPS. We tried to evaluate opendiameter project for this (http://www.opendiameter.org/) Regards, Bogdan Thank you very much. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Diameter
this looks a very good starting point, this module is designed to work with SER. I'll try to check it. Thank you. Regards On Tue, 2009-09-22 at 19:26 +0200, Stefan Sayer wrote: Hi, o Ghaith ALKAYYEM [09/21/09 19:59]: Hello lists, I'm interested in AAA functions according to Diameter which is newer than Radius. There's a module in OpenSIPS which is called AUTH_DIAMETER Module and it's mentioned that this module is obsolete. So I'd like your recommendations about this matter, should I work from the scratch to develop something that does this functionalities or is it possible to integrate other open source software with OpenSIPS. also have a look at cdp module from openimscore (http://www.openimscore.org/docs/ser_ims/CDP.html), its based on the old disc implementation, but extended a lot from that point - chances are you can use/reuse many things from there. hth Stefan Thank you very much. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Diameter
Hello lists, I'm interested in AAA functions according to Diameter which is newer than Radius. There's a module in OpenSIPS which is called AUTH_DIAMETER Module and it's mentioned that this module is obsolete. So I'd like your recommendations about this matter, should I work from the scratch to develop something that does this functionalities or is it possible to integrate other open source software with OpenSIPS. Thank you very much. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips vs kamailio
Hello, Could you please tell me the difference between these two products: OpenSIPS Kamailio every time I check them I feel they are very similar. Regards. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed
Does the dispatcher work on the same machine also? Could you provide me with more details about running these two instances? Regards. On Thu, 2009-09-10 at 23:39 +0800, Jiang Jinke wrote: Thanks for the detail instruction. I just use a symlink into the directory, it's working properly now. Just like below: /usr/local/relay1/media-relay - /usr/bin/media-relay /usr/local/relay2/media-relay - /usr/bin/media-relay Regards, Jinke Jiang On Thu, Sep 10, 2009 at 10:02 PM, Dan Pascu d...@ag-projects.com wrote: On 10 Sep 2009, at 15:47, Raúl Alexis Betancor Santana wrote: On Thursday 10 September 2009 11:56:00 Ghaith ALKAYYEM wrote: Hello, I think it's not possible to use two separate relays on the same server, I tried that a lot then I switched to RTPproxy. That's not true, you could run as many Realys as you want on the same server, only have to patch mediaproxy-relay to be able to call it with a diferent .cfg as the default one, have diferent listen ports and no more. You don't need to patch anything. Just unpack mediaproxy in as many different directories as you need, run ./build_inplace and modify each config.ini in those directories as needed. Then run mediaproxy from those directories and each of them will use the local config.ini from its own directory. Alternatively, if you want to use a system wide installation, you can copy the binaries from /usr/bin to a number of different directories and add a config.ini in each directory. Then run those binaries from those directories instead of /usr/bin/ and each binary will use the config.ini file in its own directory to overwrite settings from the global /etc/mediaproxy/config.ini. Mediaproxy uses 2 configuration files. The global one resides in /etc/ mediaproxy/config.ini. On top of that if a config.ini is present in the same directory as the binary (media-relay media-dispatcher) that one will be used to overwrite the settings from the global one having priority over it. -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] questions about log?
http://www.opensips.org/Resources/DocsTsStart On Wed, 2009-09-09 at 08:04 +0200, Uwe Kastens wrote: Hi, you can define the syslog facility in opensips.cfg. After that you can put the log to any location. BR Uwe ASHWINI NAIDU schrieb: By default the logging of opensips will be done in */var/log/syslog* in debian systems and */var/log/messages* in redhat based systems 2009/9/9 zhangchao1 zhangchao...@163.com mailto:zhangchao...@163.com Hello everybody, dose anyone know where the log file is? 中国制造,讲述中国60年往事 http://news.163.com/madeinchina/index.html?from=mailfooter ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Checksum incorrect maybe caused by UDP checksum offload
Hello, Could anybody tell me what is the problem with such a configuration: --- (UAC)192.168.10.1 |=== |192.168.10.10 [OpenSIPS] 192.168.20.20|=== |(UAC)192.168.20.1 | --- The mysterious point is that all packets orginating from the interface (192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP level. This happened with/without RTPproxy, so i'm wondering whether I should care about this problem or not because the sound is conveyed and everything seems okay. Regards. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Checksum incorrect maybe caused by UDP checksum offload
I'm using Wireshark Saúl Ibarra sag...@gmail.com a écrit : How are you making the packet captures? If tcpdump, did you use -s0? -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Development SDP header
Hello, I'm trying to investigate the media type inside the SDP header, when I had a look at the structure of SDP inside sdp.h, I found that: sdp_info contains a list of sessions and each session is constituted of more than one stream which contains the media header field. So, how can I reach the real type of media? Thank you very much. Regards. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] IDE
What debugger for development do you recommend as well? Regards. On Thu, 2009-09-03 at 09:21 +0200, Saúl Ibarra wrote: What about this? http://www.vim.org/scripts/script.php?script_id=2242 It's not very updated, but after reading it a bit seems easy to add the new core parameters and functions so that it works with latest version of OpenSIPS. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] struct sip_msg
Hello list,
I was trying to play with the SIP header, So when i tried to access the
fields (to,from) in the sip_msg structure through a module c function
they were NULL and everything was included in the field headers.
I'd like to know whether there's something wrong or it's natural for
these fields to be NULL.
struct sip_msg {
unsigned int id; /* message id, unique/process*/
struct msg_start first_line; /* Message first line */
struct via_body* via1; /* The first via */
struct via_body* via2; /* The second via */
struct hdr_field* headers; /* All the parsed headers*/
struct hdr_field* last_header; /* Pointer to the last parsed header*/
hdr_flags_t parsed_flag; /* Already parsed header field types */
/* Via, To, CSeq, Call-Id, From, end of header*/
/* pointers to the first occurrences of these headers;
* everything is also saved in 'headers' (see above)
*/
/* shorcuts to known headers */
struct hdr_field* h_via1;
struct hdr_field* h_via2;
struct hdr_field* callid;
struct hdr_field* to;
struct hdr_field* cseq;
struct hdr_field* from;
...
...
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Re: [OpenSIPS-Users] struct sip_msg
Thank you very much for these valuable information, actually I'm trying
to learn how to develop new module, and i expected that those fields
should be filled upon the receipt of any external message.
Regards.
On Wed, 2009-09-02 at 15:37 +0300, Anca Vamanu wrote:
Hi Ghaith,
You must explicitly call parse_headers for the message to be parsed and
for the fields in sip_msg to be filled.
Example:
parse_headers(msg,HDR_EOH_F, 0); - will parse all headers.
However beware that this function will not parse the headers value also.
You must call the parse function for that header and it will fill the
'parsed' filed of the struct hdr_field with a structure specific for
that filed
Example: If you want to parse the value of the Contact header filed, you
call
parse_contact(msg-contact).
and it will fill the parsed filed with a contact_body_t structure that
contains the parsed value of the Contact header.
(contact_body_t* )msg-contact-parsed;
I suggest learning by example technique :), look in other modules that
use the parser and see how it is done there. One option is function
*extract_sdialog_info* function from modules/presence/subscribe.c file.
Regards,
Anca
Ghaith ALKAYYEM wrote:
Hello list,
I was trying to play with the SIP header, So when i tried to access the
fields (to,from) in the sip_msg structure through a module c function
they were NULL and everything was included in the field headers.
I'd like to know whether there's something wrong or it's natural for
these fields to be NULL.
struct sip_msg {
unsigned int id; /* message id, unique/process*/
struct msg_start first_line; /* Message first line */
struct via_body* via1; /* The first via */
struct via_body* via2; /* The second via */
struct hdr_field* headers; /* All the parsed headers*/
struct hdr_field* last_header; /* Pointer to the last parsed header*/
hdr_flags_t parsed_flag; /* Already parsed header field types */
/* Via, To, CSeq, Call-Id, From, end of header*/
/* pointers to the first occurrences of these headers;
* everything is also saved in 'headers' (see above)
*/
/* shorcuts to known headers */
struct hdr_field* h_via1;
struct hdr_field* h_via2;
struct hdr_field* callid;
struct hdr_field* to;
struct hdr_field* cseq;
struct hdr_field* from;
...
...
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[OpenSIPS-Users] IDE
Hello, Could anybody suggest an IDE that will ease the development debugging of OpenSIPS? Thank you ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Hi, Is it possible also to make bridging dependent on a variable value by passing a variable as a parameter to force_send_socket() as following: $var(a) = x.x.x.x:xx; force_send_socket($var(a)); because the above configuration gave me an error but when I used the variable in xlog function it was okay: xlog($var(a)); I might do some code modification in this regard. Regards. On Mon, 2009-08-24 at 18:03 +0300, Bogdan-Andrei Iancu wrote: Hi Matthew, There 2 things when comes bridging: 1) signalling part - selecting the proper outbound interface (private or public) a) this can be automatically done by opensips (based on the destination IP) if you enable the mhomed parameter in core ; this is simple by not so efficient b) you can do it manually, by selecting from script the correct interface - see the force_send_socket() function 2) media part a) rtpproxy - when enabling RTPproxy (at request and reply time) you can explicitly select which interface to use (see the e and i flags - http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id271362) Best regards, Bogdan Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
It's possible also to use RTPproxy, because it's designed to work in bridging mode where it can forward the traffic from the internal network towards external one. Regards On Thu, 2009-08-20 at 12:55 -0600, Darren Sessions wrote: Media Proxy will work with public internet addresses but, and this is just my understanding, was not built for use with private IPs let alone bridging from one subnet to another. If your keeping the gateways on a private network is for security purposes, you may consider giving them public ips on the same subnet as your opensips and mediaproxy setup, but not specifying a default gateway. Essentially, this would allow the media proxy to do its job relaying the audio, while still preventing 99% of any unwanted traffic to your gateways. Couple that will firehol or some other cool iptables app (or manually configure it if you like) and you'd be sitting pretty secure I would think. Really depends on what you've designed (and why). - Darren On Aug 20, 2009, at 12:49 PM, Matthew S. Crocker wrote: I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket). Will it perform the functions to proxy the SIP RTP streams (via mediaproxy) between my end users and my internal gateway? At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc. -Matt - Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Matthew S. Crocker President Crocker Communications, Inc. PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com P: 413-746-2760 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Hello, I tried to use mediaproxy, it includes two softwares (dispatcher relay), I tried a lot to run more than one relay on the same server in order to bind them to different interfaces. But unfortunately this didn't work and I think it's not possible. I recommend using RTPProxy which is designed to work in bridging mode between two networks and you can run multiple instance of RTPProxy on the same server. Regards. On Thu, 2009-08-20 at 15:48 -0400, Matthew S. Crocker wrote: Can mediaproxy glue two RTP streams together (CallerA to CallerB)? Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 eth1) ? If so then it should be able to glue two calls together between public IP (eth0) and private IP (eth1). If the two RTP streams have to be on the same interface for mediaproxy to work then I would expect to run into issues. EndUser - (eth0) MediaProxy (eth1) - SIP Gateway - Jeff Pyle jp...@fidelityvoice.com wrote: From: Jeff Pyle jp...@fidelityvoice.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 2:52:35 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, While I'm no Mediaproxy expert, I have seen many conversations on this list where Mediaproxy is described as a part of a far-end NAT solution. It was not designed to have a private IP attached to it. For that, you most likely will want to look at the rtpproxy application. It sounds like you are constructing a local ALG to connect private and public networks. You don't necessarily need a full-blown Acme for that. I've had great luck with Edgewater Networks' Edgemarc devices, for example. That's just one. There are many. - Jeff On 8/20/09 2:49 PM, Matthew S. Crocker matt...@corp.crocker.com wrote: I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket). Will it perform the functions to proxy the SIP RTP streams (via mediaproxy) between my end users and my internal gateway? At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc. -Matt - Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] AVP/ Programing Documentation
Hi,, I think this book is available through rapidshare in the following link: http://rs211.rapidshare.com/files/143531736/Building_Telephony_Systems_with_Openser.pdf On Wed, 2009-08-19 at 11:26 -0500, osiris123d wrote: Also a good idea is to read Flavio E. Goncalves book http://www.packtpub.com/building-telephony-systems-with-openser/book Building Telephony systems with OpenSER . Though the book does deal with an old version of OpenSER it still has good documentation and examples. There is also a section on AVPOPS roger wilbert wrote: Can anyone point me to documentation other than the module docs that can explain how to AVP? Not that the module docs don’t provide good information. But it assumes knowledge that is missing from someone who is not readily familiar with Opensips. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] AVP/ Programing Documentation
I totally agree with you. I'm really sorry for that. On Wed, 2009-08-19 at 12:11 -0500, Khan wrote: Ghaith, That would be violation of copyright law won't it or else we wont buy this book for $40+ Authour has putin hardwork in putting it together, dont you think he should benefit a little ? After all OpenSIPS is free but not everything else lol... On Wed, Aug 19, 2009 at 11:41 AM, Ghaith ALKAYYEMghaith.alkay...@telecom-bretagne.eu wrote: Hi,, I think this book is available through rapidshare in the following link: On Wed, 2009-08-19 at 11:26 -0500, osiris123d wrote: Also a good idea is to read Flavio E. Goncalves book http://www.packtpub.com/building-telephony-systems-with-openser/book Building Telephony systems with OpenSER . Though the book does deal with an old version of OpenSER it still has good documentation and examples. There is also a section on AVPOPS roger wilbert wrote: Can anyone point me to documentation other than the module docs that can explain how to AVP? Not that the module docs don’t provide good information. But it assumes knowledge that is missing from someone who is not readily familiar with Opensips. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] mediaproxy relay
Hi, Could you tell me what is happening to mediaproxy relay when I'm trying to run it? The relay is not starting and this message appears: Set resource limit for maximum open file descriptors to 11000 debug: Adding new dispatcher at O j: operation is not possible without initialized secure memory Aborted ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] media proxy
Hi, Is it possible to define multiple relays on the same server that runs Opensips + Mediaproxy. I tried to define two interfaces in the configuration file, like: relay_ip = 1.1.1.1 2.2.2.2 or relay_ip = 1.1.1.1 relay_ip = 2.2.2.2 and I tried also to force the traffic to go through specific interface by calling: $avp(s:media_relay) = 1.1.1.1; and the traffic goes always through one interface. Regards. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] PROBLEM: Opensips stops replying to SIP packets
Yeah, I agree with you this also happened with me in less than one day of continuous working. On Sat, 2009-08-08 at 19:59 -0700, James Lamanna wrote: Hi, I'm running the svn 1.5 branch of opensips. I've noticed after some amount of time, usually a day or so, opensips just completely stops responding to incoming SIP requests, REGISTER, NOTIFY, etc... The only way to recover from this is to restart opensips. In fact it seems to stop doing anything, including writing debug output. Thank you. -- JAmes ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] load balancing in opensips
Hi,
I'm trying to connect OpenSIP to 3 WAN links, and I want to distribute
the traffic between those links according to the dialing number. So, I
tried some modules like lcr and load_balance but the test didn't succeed
because those modules are forwarding the traffic to another SIP gateway
or media gateway.
I tried to change the outbound interface via force_send_socket and
setting mhomed=1 and OpenSIPS still connect through the main interface.
The IPs of the system are:
172.23.1.20
172.23.1.19
172.23.1.18
and the originating IP is always 172.23.1.20
please find a part of my configuration file:
# main request routing logic
mhomed=1
route{
if($ct=~2000.*)
{
#load_balance(2,conf);
force_send_socket(172.23.1.19:5060);
};
if($ct=~3000.*)
{
#load_balance(1,conf);
force_send_socket(172.23.1.18:5060);
};
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction
fails
} else if (is_method(INVITE)) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy clients
do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(1);
} else {
/* uncomment the following lines if you want to
enable presence */
##if (is_method(SUBSCRIBE) $rd ==
your.server.ip.address) {
## # in-dialog subscribe requests
## route(2);
## exit;
##}
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
t_relay();
exit;
} else {
# ACK without matching
transaction -
# ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
#initial requests
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authenticate if from local subscriber (uncomment to enable
auth)
# authenticate all initial non-REGISTER request that pretend to
be
# preloaded route checking
if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();
# account only INVITEs
if (is_method(INVITE)) {
setflag(1); # do accounting
}
if (!uri==myself)
## replace with following line if multi-domain support is used
##if (!is_uri_host_local())
{
append_hf(P-hint: outbound\r\n);
# if you have some interdomain connections via TLS
##if($rd==tls_domain1.net) {
## t_relay(tls:domain1.net);
## exit;
##} else if($rd==tls_domain2.net) {
## t_relay(tls:domain2.net);
## exit;
##}
route(1);
}
# requests for my domain
## uncomment this if you want to enable presence server
## and comment the next 'if' block
## NOTE: uncomment also the definition of route[2] from below
##if( is_method(PUBLISH|SUBSCRIBE))
## route(2);
if (is_method(PUBLISH))
{
sl_send_reply(503, Service Unavailable);
exit;
}
if (is_method(REGISTER))
{
# authenticate the REGISTER requests (uncomment to enable
auth)
##if (!www_authorize(,
Re: [OpenSIPS-Users] Trouble with perl module
Hi; The same thing happened to me, But when i tried to call the script through the configuration file, it was okay and executed successfully. On Mon, 2009-07-13 at 13:55 +0400, M C wrote: Hello, I ve installed perl module. Also, i copied directory with perl lib OpenSIPS.pm to /usr/lib/perl. But when i am trying to execute a simple code: use OpenSIPS::Message; print Test\n; or some of examples, line functions.pl, i have error: Can't locate object method bootstrap via package OpenSIPS at /usr/lib/perl/5.10/OpenSIPS/ Message.pm line 32. How can i fix it? -- Best regards, Maksim. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] perl module
Hi guys, I'm trying to use the perl modules available with OpenSIPS, but after installing it I encountered a problem when i tried to compile a script: Can't locate object method bootstrap via package OpenSIPS at /usr/lib/perl5/OpenSIPS/Message.pm line 32. Compilation failed in require at /usr/lib/perl5/OpenSIPS.pm line 34. BEGIN failed--compilation aborted at /usr/lib/perl5/OpenSIPS.pm line 34. Compilation failed in require at lib/opensips/perl/functions.pl line 4. BEGIN failed--compilation aborted at lib/opensips/perl/functions.pl line 4. does anybody have any idea about this issue? Thanks in advance. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
