Re: [OpenSIPS-Devel] [ opensips-Bugs-3189178 ] b2b_entites: control of the validity of CANCEL
Could you backport it to the 1.6.x branch please ?
Regards,
On Tue, Feb 22, 2011 at 2:05 PM, SourceForge.net
nore...@sourceforge.net wrote:
Bugs item #3189178, was opened at 2011-02-22 07:30
Message generated for change (Comment added) made by osas
You can respond by visiting:
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Please note that this message will contain a full copy of the comment thread,
including the initial issue submission, for this request,
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Category: modules
Group: 1.6.x
Status: Open
Resolution: None
Priority: 5
Private: No
Submitted By: Nobody/Anonymous (nobody)
Assigned to: Nobody/Anonymous (nobody)
Summary: b2b_entites: control of the validity of CANCEL
Initial Comment:
Hi,
in prescript function in dlg.c line 543, you check if CallID looks like B2B
hash key to prevent self sending.
543 if(b2b_parse_key(callid, hash_index, local_index)
= 0)
544 {
545 LM_DBG(received a CANCEL message that I
sent\n);
546 return 1;
547 }
But if there are linked b2b, and a node Cancel a session, the cancel is not
send to logic because CallID looks like B2B hash.
Regards,
--
Comment By: Ovidiu Sas (osas)
Date: 2011-02-22 08:05
Message:
If you are chaining several opensipsnservers running in b2b mode, then use
different prefixes:
http://www.opensips.org/html/docs/modules/devel/b2b_entities.html#id250032
Regards,
Ovidiu Sas
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Re: [OpenSIPS-Devel] [ opensips-Bugs-3178281 ] Srand in b2b_entities
Here is my patch. Why: srand has to be initiate with a different value by process. In the previous version, srand called get_uticks with concurrency access, and rand value will be the same. On Fri, Feb 11, 2011 at 2:21 PM, SourceForge.net nore...@sourceforge.net wrote: Bugs item #3178281, was opened at 2011-02-11 13:21 Message generated for change (Tracker Item Submitted) made by nobody You can respond by visiting: https://sourceforge.net/tracker/?func=detailatid=1086410aid=3178281group_id=232389 Please note that this message will contain a full copy of the comment thread, including the initial issue submission, for this request, not just the latest update. Category: modules Group: 1.6.x Status: Open Resolution: None Priority: 5 Private: No Submitted By: Nobody/Anonymous (nobody) Assigned to: Nobody/Anonymous (nobody) Summary: Srand in b2b_entities Initial Comment: Hi, The use of srand is wrong in generation of random value in the second part of callid. srand have to be called one time for init, after call rand to get a random value. Regards, -- You can respond by visiting: https://sourceforge.net/tracker/?func=detailatid=1086410aid=3178281group_id=232389 ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel -- Olivier Détour srand.patch Description: Binary data ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [B2B_ENTITES] SIP OPTION Support
Hi, I would like to know if SIP OPTION is supported in the b2b_entitites ? And if a support is possible ? Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [B2B_ENTITES] SIP OPTION Support
On Fri, Jan 21, 2011 at 5:37 PM, Anca Vamanu a...@opensips.org wrote: On 01/21/2011 06:28 PM, Olivier Détour wrote: Hi, I would like to know if SIP OPTION is supported in the b2b_entitites ? And if a support is possible ? You mean if the B2B is able to send Option pings to call end users? Send and Receive it. Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [B2B_ENTITES] Patch port checking in prescript
Hi, here is my patch to complete checking if there is no port in configuration or RURI. Regards, -- Olivier Détour port_b2b_entites.patch Description: Binary data ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] About b2b_entities
Oh thank you very much. And about my other question: will it be possible to disable db features in b2b_entities via cfg ? Regards, On Wed, Nov 17, 2010 at 3:17 PM, Ovidiu Sas o...@voipembedded.com wrote: It seems that you are using sipp to generate this scenario. In the ACK, you have a different branch then in the original INVITE. Try to use a real client to test the b2b module. Regards, Ovidiu Sas 2010/11/17 Olivier Détour chino540off+kamai...@gmail.com: Hi, Do you have any news about my problem ? Thanks, 2010/11/4 Olivier Détour chino540off+kamai...@gmail.com: Here is my PCAP. On Thu, Nov 4, 2010 at 12:18 PM, Anca Vamanu a...@opensips.org wrote: Hi Olivier, This means that the ACK is not matched with the transaction. Can you send a network trace with the 3 packages? Maybe there is something wrong with the ACK. Regards, -- Anca Vamanu www.voice-system.ro On 11/03/2010 07:08 PM, Olivier Détour wrote: Hi, I found another bug today: If I create an UAS and reject call, the 4XX Reply is resent. I release my UAS after the ACK. Caller INVITE--- B2B ---4XX ACK--- -- 4XX--- -- 4XX--- -- 4XX--- -- 4XX--- Regards, ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel -- Olivier Détour -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] About b2b_entities
Hi, Do you have any news about my problem ? Thanks, 2010/11/4 Olivier Détour chino540off+kamai...@gmail.com: Here is my PCAP. On Thu, Nov 4, 2010 at 12:18 PM, Anca Vamanu a...@opensips.org wrote: Hi Olivier, This means that the ACK is not matched with the transaction. Can you send a network trace with the 3 packages? Maybe there is something wrong with the ACK. Regards, -- Anca Vamanu www.voice-system.ro On 11/03/2010 07:08 PM, Olivier Détour wrote: Hi, I found another bug today: If I create an UAS and reject call, the 4XX Reply is resent. I release my UAS after the ACK. Caller INVITE--- B2B ---4XX ACK--- -- 4XX--- -- 4XX--- -- 4XX--- -- 4XX--- Regards, ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel -- Olivier Détour -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] About b2b_entities
Here is my PCAP. On Thu, Nov 4, 2010 at 12:18 PM, Anca Vamanu a...@opensips.org wrote: Hi Olivier, This means that the ACK is not matched with the transaction. Can you send a network trace with the 3 packages? Maybe there is something wrong with the ACK. Regards, -- Anca Vamanu www.voice-system.ro On 11/03/2010 07:08 PM, Olivier Détour wrote: Hi, I found another bug today: If I create an UAS and reject call, the 4XX Reply is resent. I release my UAS after the ACK. Caller INVITE--- B2B ---4XX ACK--- -- 4XX--- -- 4XX--- -- 4XX--- -- 4XX--- Regards, ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel -- Olivier Détour 404_not_found.pcap Description: Binary data ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] About b2b_entities
Hi, I'm using v1.6 from SVN, and I would like to report an UPDATE in my b2b client module, but entities reply on INVITE with 491 error code. Here is my test (from RFC 3311): Caller INVITE--- B2B ---180 --UPDATE-- -- 491--- In sources, this reply happen when the call state is proceeding, but I don't think it is RFC compliant ? I saw you added DB binding in entities, but this feature is set by default. Would you make it configurable (use it or not) ? Thanks you, Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] Problem with UAC in b2b_entities and tcp connections
Hi, I would like to understand something: I am writing a b2b module. My clients are only over TCP/TLS and the connection is already created at this time. When I created a UAC with b2b_entities, I can get tcp informations (ip/port), and I format my request URI to be parsed by the research function to get tcp_connection context. Fine. But after a reply on my INVITE, I would like to ACK it, but b2b_entites will use Contact Header to reply (cf RFC), but your research function cannot find context because it use wrong informations. So my question is: how could we do this research TCP informations with SIP informations ? Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [B2B] How to create a UAC with TLS
On Wed, May 12, 2010 at 12:56 PM, Anca Vamanu a...@opensips.org wrote: Olivier Détour wrote: 2010/5/10 Olivier Détour chino540off+kamai...@gmail.com: Hi, I would like to create an UAC with client_new over a TLS connection. My peer is registered and the TLS tunnel is set up. But when I create my UAC, and send my first request to my peer, the sending failed. I saw I have to give a sock_info* in client_info_t*. But how do I find the right socket_info to send my request on the right tunnel ? Thanks for your replies, Regards, After more investigation, how could I get tcp_connect id to msg_send from my module ? Does it possible or not ? Regards, Hi Olivier, Putting the same RURI as the contact in the Register sent by client does not work? From what I know, opensips will search through its tcp connections for a match of the contact and use that tunnel. Regards, I found a dirty solution: get tcp port from REGISTER and inject it in TO uri in client_new. It's working, Thanks you, Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [B2B] How to create a UAC with TLS
2010/5/10 Olivier Détour chino540off+kamai...@gmail.com: Hi, I would like to create an UAC with client_new over a TLS connection. My peer is registered and the TLS tunnel is set up. But when I create my UAC, and send my first request to my peer, the sending failed. I saw I have to give a sock_info* in client_info_t*. But how do I find the right socket_info to send my request on the right tunnel ? Thanks for your replies, Regards, After more investigation, how could I get tcp_connect id to msg_send from my module ? Does it possible or not ? Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [B2B] How to create a UAC with TLS
Hi, I would like to create an UAC with client_new over a TLS connection. My peer is registered and the TLS tunnel is set up. But when I create my UAC, and send my first request to my peer, the sending failed. I saw I have to give a sock_info* in client_info_t*. But how do I find the right socket_info to send my request on the right tunnel ? Thanks for your replies, Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [B2B_ENTITES] - new_server/client.
On Thu, Apr 29, 2010 at 5:21 PM, Anca Vamanu a...@opensips.org wrote: Olivier Détour wrote: Hi, I would like to understand why param when I want to create a new UAS in my B2B_module has to be str*. Why don't you use a simple void* in struct b2b_dlg to give context to callbacks ?? There is something I don't understand with this features ?? (If yes, you have to check if param == NULL in b2b_new_dlg before to copy it line 795: dlg.param = *param;) Regards, Hi Olivier , Initial the param was void*, then it became str* when I wanted to do db storage also ( it is still not completed, but will be soon). I do the change that you suggested - indeed it was not safe. Regards, Thanks you, Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [B2B_ENTITES] - new_server/client.
Hi, I would like to understand why param when I want to create a new UAS in my B2B_module has to be str*. Why don't you use a simple void* in struct b2b_dlg to give context to callbacks ?? There is something I don't understand with this features ?? (If yes, you have to check if param == NULL in b2b_new_dlg before to copy it line 795: dlg.param = *param;) Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [B2B_ENTITIES] Difference between sending CANCEL and BYE
2010/1/28 Anca Vamanu a...@opensips.org: Hi Olivier, First, please do not send email to me privately ( use reply all). Sorry about that, I did not see CC list, You are saying that the CANCEL send by b2b_entities does not have the exact ruri as INVITE? No I'm saying exactly the opposite, I know sip's RFC about CANCEL construction with INVITE header. But there a case, if B2B_entities communicates with another one, CANCEL is correctly sent to the second B2B_entities. Did you make some tests with 2 B2B face to face and try to: - CANCEL from a phone and propagate it correctly; - CANCEL from a B2B and propagate it to the callee phone; in this case, what is the best way: Send CANCEL on a side and send a 487 on the other side; or send CANCEL and wait 487 from the callee phone and propagate it to the caller phone ? Can you catch an INVITE and the corresponding CANCEL (send by b2b_entities) and send them in an e-mail? I don't have time to send it before after next week. I will come back to you after that and I will give you more information about SIP transmission, and what I am expected for my project. Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [B2B_ENTITIES] Difference between sending CANCEL and BYE
2010/1/22 Olivier Détour chino540off+kamai...@gmail.com:
Hi,
I understand the problem, but I would like to explain you my topology:
Broker1 Broker2
| |
Tel(10) -- B2BUA1 -- B2BUA2 -- Tel(20)
My problem is when 10 calls 20, 20 is ringing (so B2BUAs create their
UAS and UAC).
I want to be able to brake the current SIP session at this moment.
So I send the CANCEL to UAC side on B2BUA1. But B2BUA2 doesn't want to
receive it,
because It is not send to sip:s...@1.1.12.3. So message could not be
transmit to B2BUA2 ...
Maybe, I don't understand something here, but b2b_entities API could
be able to manage
this situation ?
Regards,
Hi,
I went through the source code of B2B_entities, and I don't understand
when you search the dlg in int b2b_prescript_f(struct sip_msg *msg,
void *uparam) in dlg.c:
357 while(dlg)
358 {
359 if(ruri.len == dlg-ruri.len strncmp(ruri.s,
dlg-ruri.s,
ruri.len)== 0
360dlg-callid.len == callid.len
361 strncmp(dlg-callid.s, callid.s,
callid.len)== 0
362 dlg-tag[CALLER_LEG].len ==
from_tag.len
363 strncmp(dlg-tag[CALLER_LEG].s,
from_tag.s, from_tag.len)== 0)
You check CALLID, FROM_TAG and the REQUEST_URI. If you create the UAS
with B2BUA_entities,
Requests send by caller will change Request URI to speaks directly with UAS.
Why is the RURI in your search?
When I remove the RURI check, it works as my understanding of the RFC.
Regards,
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[OpenSIPS-Devel] [B2B_ENTITIES] bug in prescript
Hi,
There is a bug in prescript function when you compare RURI with server_addr:
296 if (method_value!= METHOD_CANCEL
297 !((msg-first_line.u.request.uri.len ==
server_address.len )
298 strncmp(msg-first_line.u.request.uri.s, server_address.s,
299 server_address.len)== 0))
300 {
301 LM_DBG(RURI does not point to me\n);
302 return 1;
303 }
with opensips.cfg:
modparam(b2b_entities, server_address, sip:s...@10.10.10.10)
Imagine if the peer change RURI with adding the default port at the
end of the string,
or remove it: the strncmp is not sufficient.
I am working with a Cisco, and it happens.
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Re: [OpenSIPS-Devel] [B2B_ENTITIES] ACK Strategy
2010/1/21 Olivier Détour chino540off+kamai...@gmail.com: Hi, I'm having a hard time using b2b_entities: After sending a 4xx error (long after the invite), I'd like to be notified when the client sends the ACK. I'd want to be able to close the call context when my module receives the client's ACK, but I never see it, b2b_entities silently processes it. Is it possible to change this behaviour? I saw in the debug trace (debug=9) that the ACK is not for the server_address parameter (in opensips.cfg), could this be a part of the problem? For example I've got a 404 error being resent continuously until phone timeout. If I missed documentation on the subject I will gladly RTFM if someone points me in the right direction. Thanks, Regards, Sorry about that, configuration problems, thanks, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [B2B_ENTITIES] Difference between sending CANCEL and BYE
Hi, I would like to break a SIP communication during a call or a proceeding SIP session... If the call is in progress (Caller can speak to Callee), I can send a BYE to caller and callee each with send_request function; Here is my Wireshark trace: BYE sip:s...@1.1.12.3 SIP/2.0 Via: SIP/2.0/UDP 2.2.22.2;branch=z9hG4bKda73.5742b3a7.0 From: sip:1...@2.2.22.2;tag=934c0604001f8a49c065a1707fbb682d-59cf To: sip:2...@1.1.12.3;tag=B2B.333.0.1264181639 CSeq: 4 BYE Call-ID: B2B.269.0.1264181639 Content-Length: 0 User-Agent: OpenSIPS (1.6.1-notls (i386/linux)) Contact: sip:s...@2.2.22.2 If the sip session is proceeding (Caller send INVITE but he hasn't receive 200 OK yet), I send a CANCEL like I sent the BYE (send_request); But here is my Wireshark trace: CANCEL sip:2...@1.1.12.3 SIP/2.0 Via: SIP/2.0/UDP 2.2.22.2;branch=z9hG4bK5161.0ab21263.0 From: sip:1...@2.2.22.2;tag=934c0604001f8a49c065a1707fbb682d-ac1f To: sip:2...@1.1.12.3 CSeq: 2 CANCEL Call-ID: B2B.134.0.1264181129 User-Agent: OpenSIPS (1.6.1-notls (i386/linux)) Contact: sip:s...@2.2.22.2 Why is the Request Line not for sip:s...@1.1.12.3 as for the BYE? Regards, -- Olivier Détour Sent from Paris, Île-de-France, France ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [B2B_ENTITIES] Difference between sending CANCEL and BYE
Hi, I understand the problem, but I would like to explain you my topology: Broker1 Broker2 | | Tel(10) -- B2BUA1 -- B2BUA2 -- Tel(20) My problem is when 10 calls 20, 20 is ringing (so B2BUAs create their UAS and UAC). I want to be able to brake the current SIP session at this moment. So I send the CANCEL to UAC side on B2BUA1. But B2BUA2 doesn't want to receive it, because It is not send to sip:s...@1.1.12.3. So message could not be transmit to B2BUA2 ... Maybe, I don't understand something here, but b2b_entities API could be able to manage this situation ? Regards, -- Olivier Détour Sent from Paris, Ile-de-France, France ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] [B2B_ENTITIES] ACK Strategy
Hi, I'm having a hard time using b2b_entities: After sending a 4xx error (long after the invite), I'd like to be notified when the client sends the ACK. I'd want to be able to close the call context when my module receives the client's ACK, but I never see it, b2b_entities silently processes it. Is it possible to change this behaviour? I saw in the debug trace (debug=9) that the ACK is not for the server_address parameter (in opensips.cfg), could this be a part of the problem? For example I've got a 404 error being resent continuously until phone timeout. If I missed documentation on the subject I will gladly RTFM if someone points me in the right direction. Thanks, Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [ opensips-Bugs-2932845 ] 180 Ringing not forward to B2B_logic
2010/1/15 Olivier Détour chino540off+kamai...@gmail.com: On Fri, Jan 15, 2010 at 2:51 PM, SourceForge.net nore...@sourceforge.net wrote: Bugs item #2932845, was opened at 2010-01-15 14:49 Message generated for change (Settings changed) made by anca_vamanu You can respond by visiting: https://sourceforge.net/tracker/?func=detailatid=1086410aid=2932845group_id=232389 Please note that this message will contain a full copy of the comment thread, including the initial issue submission, for this request, not just the latest update. Category: modules Group: 1.6.x Status: Open Resolution: None Priority: 5 Private: No Submitted By: Nobody/Anonymous (nobody) Assigned to: Anca Vamanu (anca_vamanu) Summary: 180 Ringing not forward to B2B_logic Initial Comment: When a 180 ringing comes on OpenSIPs, it is not forward to B2B_logic module. -- Comment By: Anca Vamanu (anca_vamanu) Date: 2010-01-15 15:51 Message: Hi, Have you set the module parameter in tm: modparam(tm, pass_provisional_replies, 1) ? Yes, this parameter is set. Regards, Anca As the precedent problem, why ACK request is not forward to the client module ? I would like to get ACK after sending a 4xx reply to remove my context. If I remove it after sending 4xx, ACK is received by B2B_entities but there is no dialogue for my key. How I could do something ? Regards, -- Olivier Détour Sent from Paris, Île-de-France, France ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] Share SIP Messages between 2 processes
On Fri, Jan 15, 2010 at 12:52 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Olivier Détour wrote: On Thu, Jan 14, 2010 at 6:55 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Olivier, Olivier Détour wrote: Hi, I'm writing a B2B module for the B2B_entities module (like B2B_logic). In my initialisation, I create an extra process to poll on an extra FD to receive order to create UAC part. Why don't you use the MI stuff for this? you can trigger an action / event by sending an MI command to the server - you module just has to export a new MI function. I would like to know how to share a SIP message between the module and the extra process (I'm using OpenSIPs in fork mode ...) ? I tried to shm_malloc it but I get an out of memory error on the second Communication. normally using shm_malloc is the trick, but depends of where you fork your process (if the shm is inherited) and who you use it for transferring data from from A to B. Forks come from OpenSIPs itself (worker forks + my extra fork (extra process in module declaration)). OK, that's correct I think I have to use shared memory to share data between different processes. I tried to find the best and the most optimize way to use it. yes, you should use the sh mem. But what exactly do faile for you? I mean what exactly are you doing and what step fails? is the malloc not working? or ? When I use B2B, I have to forward my INVITE in another process, that is why I have to copy it (body, URIs, ...) in my shm_malloced internal data structure. It works, but I find it ugly. Moreover, sometimes I get an internal error because I don't have enough shared memory, that's why I want to find a better way. I'm blocked by OpenSIPs' architecture. If this is the only way, I could optimize it on my side. Do you know actually, how much percent of the 32 MB of shared memory, is used by OpenSIPs ? Depends of what modules you are using - each module may have some internal data structures that are kept in shm mem. You can check this via MI interface, after starting opensips (with no ongoing traffic) : opensipsctl fifo get_statistics all Thanks for the tips, it will be helpful to debugging. Regards, Bogdan Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
Re: [OpenSIPS-Devel] [ opensips-Bugs-2932845 ] 180 Ringing not forward to B2B_logic
On Fri, Jan 15, 2010 at 2:51 PM, SourceForge.net nore...@sourceforge.net wrote: Bugs item #2932845, was opened at 2010-01-15 14:49 Message generated for change (Settings changed) made by anca_vamanu You can respond by visiting: https://sourceforge.net/tracker/?func=detailatid=1086410aid=2932845group_id=232389 Please note that this message will contain a full copy of the comment thread, including the initial issue submission, for this request, not just the latest update. Category: modules Group: 1.6.x Status: Open Resolution: None Priority: 5 Private: No Submitted By: Nobody/Anonymous (nobody) Assigned to: Anca Vamanu (anca_vamanu) Summary: 180 Ringing not forward to B2B_logic Initial Comment: When a 180 ringing comes on OpenSIPs, it is not forward to B2B_logic module. -- Comment By: Anca Vamanu (anca_vamanu) Date: 2010-01-15 15:51 Message: Hi, Have you set the module parameter in tm: modparam(tm, pass_provisional_replies, 1) ? Yes, this parameter is set. Regards, Anca -- Olivier Détour Sent from Paris, Île-de-France, France ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
[OpenSIPS-Devel] Add a B2B module with an external socket
Hi, I would like to create a new B2B module. This module has to read an external file descriptor and call a callback. I don't want to use extra process and shared memory or something like that. I have a FD from a socket. If a SIP message comes, it writes on the FD. But I don't wait for the response. I have to be notified of a response on this socket. Is it possible to add a fd and a callback in an internal polling process without a fork (I want to avoid problems of communication between original process and the child) ? Regards, -- Olivier Détour ___ Devel mailing list Devel@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
