[SR-Users] RFC: best way to create dialogs
Hi! I use the dialog module to count and limit concurrent calls per user. It worked fine with 4.1.7 but fails with 4.2.2. My config basically looks like: route{ ... dlg_manage() ... authentication (stateless replies + exit) ... t_on_reply() t_on_branch() ... t_relay() exit; } initial_cbs_inscript uses the default value 1. Using 4.1.7, after stateless reply+exit, Kamailio executes dialog callbacks and deletes the dialog: [dlg_var.c:55]: dlg_cfg_cb(): new dialog with no trasaction after config execution [dlg_hash.c:872]: dlg_unref(): unref dlg 0xb233c3a0 with 1 - 1 [dlg_hash.c:872]: dlg_unref(): unref dlg 0xb233c3a0 with 1 - 0 [dlg_hash.c:872]: dlg_unref(): ref =0 for dialog 0xb233c3a0 [dlg_hash.c:355]: destroy_dlg(): destroying dialog 0xb233c3a0 (ref 0) With 4.2.2, after stateless reply+exit, Kamailio just exits without calling any dialog callback (to destroy the dialog) I think I could work around the problem by creating the transaction later. Here are some ideas and I would be happy about your comments and best practices. - set initial_cbs_inscript to 0 - call dlg_manage() just before t_relay - do not use dlg_manage() but use the dialog flag and set the flag somewhere before t_relay Finally, do t_on_reply() and t_on_branch() already create the transaction or is it created with t_relay()? Thanks Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dialog module generates error since upgrade to 4.2.2
Nice, I have the same problem after upgrading from 4.1.7 to 4.2.2. . in my case it seems the dialog callbacks are not executed for responses ... I am still debugging . On 22.01.2015 13:25, Jan Hazenberg wrote: Hi All, I'm running into a issue with the dialog module since the upgrade from 4.2.1 to 4.2.2. I use the dialog module to add extra vars to the cdr's generated by the ACC module. I use the following config: # - dialog params - modparam(dialog, enable_stats, 1) modparam(dialog, dlg_match_mode, 1) modparam(dialog, dlg_flag, FLT_DLG) modparam(dialog, rr_param, did) modparam(dialog, wait_ack, 1) # Create dialog if (method==INVITE) { # Create Dialog dlg_manage(); # Add test dialog var $dlg_var(src_ua) = $hdr(User-Agent); } This seems to work fine on kamailio 4.2.1 but after the upgrade i see the following errors in the logs: Jan 22 13:16:57 sip /usr/sbin/kamailio[2773]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587@192.168.149.126' and tags '4017642921' '' Jan 22 13:17:01 sip /usr/sbin/kamailio[2770]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 7 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587@192.168.149.126' and tags '4017642921' '' If i'm correct bogus event 6 in state 1 indicates that a ACK whas received while the dialog still was in unconfirmed state. Should i catch the 200 OK on the INVITE to update the dialog? Thanks, Jan Hazenberg ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] RedHat 7 packages issue
Hi all, Thanks for your hard working on Kamailio! I was just trying to install kamailio 4.2.X on a RedHat 7 machine but I got checksum mismatch: I took the yum configuration from: http://rpm.kamailio.org/ [kamailio] name=RPMs for Kamailio on RHEL 7 type=rpm-md baseurl=http://rpm.kamailio.org/stable/RHEL_7/ gpgcheck=1 gpgkey=http://rpm.kamailio.org/stable/RHEL_7/repodata/repomd.xml.key enabled=1 Than started yum update $ sudo yum update Loaded plugins: amazon-id, rhui-lb kamailio | 1.2 kB 00:00:00 kamailio/primary FAILED ] 0.0 B/s |0 B --:--:-- ETA http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz: [Errno -1] Metadata file does not match checksum] 0.0 B/s |0 B --:--:-- ETA Trying other mirror. kamailio/primary | 7.4 kB 00:00:00 http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz: [Errno -1] Metadata file does not match checksum Trying other mirror. Indeed http://rpm.kamailio.org/stable/RHEL_7/repodata/repomd.xml says: data type=primary checksum type=sha4cc322a95d4a978e383903feaf8fbbe8ccc2ff9e/checksum open-checksum type=sha9fd04e59185a8e844133b834a662cae3cc0837b7/open-checksum location href=repodata/primary.xml.gz/ timestamp1413465311/timestamp size7344/size open-size70829/open-size /data while http://rpm.kamailio.org/stable/RHEL_7/repodata/primary.xml.gz I can download is: $ ls -l primary.xml.gz -rw-rw-r-- 1 alberto alberto 7562 gen 22 10:58 primary.xml.gz $ shasum primary.xml.gz e8e052970984feb0cdb16b98f05aefc516295aba primary.xml.gz Do I have to switch to other repository? Thank you, Best Regards -- Alberto Panizzo CTO - Co-Founder Amarula Solutions BV Cruquiuskade 47 Amsterdam 1018 AM NL T. +31(0)851119171 F. +31(0)204106211 www.amarulasolutions.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] topoh ACK Call-ID mismatch
On Wednesday 21 January 2015 23:52:46 Daniel-Constantin Mierla wrote: It is an ACK for a negative response, therefore it is hop-by-hop (received and discared by kamailio, then a new one is generated to the next hop). The information of direction which was detected with Route header ftag cannot be done in this case. I will try to figure out how can be solved... Am I to naive to think that these ACKs to negatives always (callid masking and whether topoh is active or not) need to use the Call-ID from the negative (in this case 488) response? Call-ID has only to be rewritten in forwarding the negative response towards the endpoint that triggered it (which is done correctly in my call trace) -- Telefoon: 088 0100 700 Sales: sa...@pocos.nl | Service: serviced...@pocos.nl http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 17097024 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] reINVITE with uac_req_send()
Hi ! I have the following simple scenario A --- Kamailio --- Asterisk --- B now i'm trying to handle Timeout reply when B is unavailable. First i suspend transaction and wait for B available. i use uac_reg_send() to reINVITE B .The problem is when B available, it receive INVITE but A doesn't receive ACK therefore the call is incomplete I guess the reason is source IP in reINVITE is Kamailio's not A's IP, so ACK can't reach A. How can i set this source IP to A's IP ? Any idea is appreciated Thank you ! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] fix_nated_contact and IPv6
Sounds like a bug. Anyway, as fix_nated_contact is not standard conform, it is better to use handle_ruri_alias() and add_contact_alias() (see the Kamailio default config file). Maybe they handle also IPv6 correct. regards Klaus On 22.01.2015 16:26, Sebastian Damm wrote: Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: sip:bob@192.168.8.132 mailto:sip%3Abob@192.168.8.132 outbound Contact: sip:bob@1234:1234:0:1234:0:0:0:2 The correct way of transforming the Contact URI would be: Contact: sip:bob@[1234:1234:0:1234:0:0:0:2] This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] fix_nated_contact and IPv6
Couple of remarks: 1) I guess the initial author know there is no nat in ipv6, so he didn't bother with. I just pushed a patch for it in master (814c08f3), if tested and reported to work ok, it can be backported 2) the contact uri example in the first email is perhaps not properly reflecting the contact uri that was generated, because it should have been with the ip address and the port. It seems to be only the ip address. If there was an omission, that's ok, because I expect the uri parsing error is due to hostpart having the port following the ipv6 address -- that requires the ipv6 between []. If the port was missing, that can be another issue, but the code shows the port is always added and it wouldn't worked at all so far without it. 3) use set_contact_alias() if use modules that need the new contact (like dialog, presence, ...) for later usage. The *contact_alias() function don't change the host/port part, they just add a new parameter, so it would have been safe with or without []. Anyhow, the code adds [] if the address is ipv6 Cheers, Daniel On 22/01/15 16:43, Klaus Darilion wrote: Sounds like a bug. Anyway, as fix_nated_contact is not standard conform, it is better to use handle_ruri_alias() and add_contact_alias() (see the Kamailio default config file). Maybe they handle also IPv6 correct. regards Klaus On 22.01.2015 16:26, Sebastian Damm wrote: Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: sip:bob@192.168.8.132 mailto:sip%3Abob@192.168.8.132 outbound Contact: sip:bob@1234:1234:0:1234:0:0:0:2 The correct way of transforming the Contact URI would be: Contact: sip:bob@[1234:1234:0:1234:0:0:0:2] This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dialog module generates error since upgrade to 4.2.2
Hi Jan! I replaced dlg_manage with setflag(dialog flag). This way the dialog is created only when a transaction is created. It solved my problems. regards Klaus On 22.01.2015 14:47, Klaus Darilion wrote: Nice, I have the same problem after upgrading from 4.1.7 to 4.2.2. . in my case it seems the dialog callbacks are not executed for responses ... I am still debugging . On 22.01.2015 13:25, Jan Hazenberg wrote: Hi All, I'm running into a issue with the dialog module since the upgrade from 4.2.1 to 4.2.2. I use the dialog module to add extra vars to the cdr's generated by the ACC module. I use the following config: # - dialog params - modparam(dialog, enable_stats, 1) modparam(dialog, dlg_match_mode, 1) modparam(dialog, dlg_flag, FLT_DLG) modparam(dialog, rr_param, did) modparam(dialog, wait_ack, 1) # Create dialog if (method==INVITE) { # Create Dialog dlg_manage(); # Add test dialog var $dlg_var(src_ua) = $hdr(User-Agent); } This seems to work fine on kamailio 4.2.1 but after the upgrade i see the following errors in the logs: Jan 22 13:16:57 sip /usr/sbin/kamailio[2773]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587@192.168.149.126' and tags '4017642921' '' Jan 22 13:17:01 sip /usr/sbin/kamailio[2770]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 7 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587@192.168.149.126' and tags '4017642921' '' If i'm correct bogus event 6 in state 1 indicates that a ACK whas received while the dialog still was in unconfirmed state. Should i catch the 200 OK on the INVITE to update the dialog? Thanks, Jan Hazenberg ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] fix_nated_contact and IPv6
Hi, I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6 and IPv4. Everything looks pretty good. Now I have a test client which sends packets via IPv6, but in contact header, SDP etc. there are still IPv4 addresses. (This comes from some converting from v4 to v6 earlier.) Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the Contact URI is wrong. Example: inbound Contact: sip:bob@192.168.8.132 outbound Contact: sip:bob@1234:1234:0:1234:0:0:0:2 The correct way of transforming the Contact URI would be: Contact: sip:bob@[1234:1234:0:1234:0:0:0:2] This incorrect Contact URI doesn't hurt first, but when the called party wants to hang up the call, it addresses the URI from Contact header, and Kamailio can't parse the Request URI: Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv [pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]: loose_route(): failed to parse Request URI Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain [domain.c:140]: is_uri_host_local(): error while parsing R-URI Is this a bug in the nathelper module? Was it never meant to handle IPv6 addresses? Or do I understand something wrong? Best Regards, Sebastian ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Trouble getting start time / call duration from dialog module
Hi Abdul I just wanted to check int. Did you have any ideas of things we could try or more information we could get you to assistance? Thanks again for your help, the input we have gotten from you and Daniel has always been invaluable. All the best. Will Ferrer Switchsoft On Fri, Jan 16, 2015 at 7:42 PM, Will Ferrer will.fer...@switchsoft.com wrote: Hi Abdul That's great. I look forward to the advice. Thanks again for the assistance and I hope you have a great weekend. All the best. Will On Fri, Jan 16, 2015 at 6:44 PM, Abdul Gafar abdul.gafar@gmail.com wrote: Hi Wiil I will be try help you //Gafar On Sat, Jan 17, 2015 at 4:43 AM, Will Ferrer will.fer...@switchsoft.com wrote: Hi Adbul Thanks, I am glad the information is useful. Do you have any thoughts on what I could I try next to get that $dlg(start_ts) value populated in the dialog as it is still not working about adding the loose_route() function? I also tried looking at the logs for any reference to dialog -- tail -f /var/log/syslog | grep dialog The only thing that came up with rtpengine talking about dialogs so no clues there. Thanks to you guys for the help. All the best. Will On Fri, Jan 16, 2015 at 1:25 AM, Abdul Gafar abdul.gafar@gmail.com wrote: *Hi Will,Thanks for sharing very useful information.* *//Gafar* On Fri, Jan 16, 2015 at 12:48 PM, Will Ferrer will.fer...@switchsoft.com wrote: Hi Daniel Thanks so much for the response and help as always. I tried changing my config to use loose route. It looks like this now: loadmodule dialog.so ... modparam(dialog, db_url, DBURL) modparam(dialog, db_mode, 1) modparam(dialog, dlg_flag, 4) modparam(dialog, dlg_match_mode, 1) ... request_route { if (is_method(INVITE) (! has_totag() ) ) { dlg_manage(); xlog (L_INFO, request_route DIALOG TEST: Dialog initiated); } if (is_method(BYE)) { #dlg_manage(); loose_route(); $var(elapsed) = ( $TV(s) - $dlg(start_ts) ); xlog (L_INFO, request_route DIALOG TEST: Completed $dlg(from_uri) to $dlg(to_uri), elapsed: $var(elapsed), now seconds: $TV(s), dlg start time: $dlg(start_ts), DLG_lifetime: $DLG_lifetime); } I now get: INFO: script: request_route DIALOG TEST: Dialog initiate INFO: script: request_route DIALOG TEST: Completed sip:willf1976t...@develop-sbc.switchsoft.com to sip:+18054515...@develop-sbc.switchsoft.com, elapsed: 1421386898, now seconds: 1421386898, dlg start time: 0, DLG_lifetime: 1421386898 So the $DLG_lifetime is being populated, but it has all the seconds since epoch time. You can also see that the $dlg(start_ts) is 0. I also tried using the dlg_manage() instead of the loose route in my test and got the same result. Any idea what might be missing? Thanks again for your help. All the best. Will On Thu, Jan 15, 2015 at 5:53 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, try to do dlg_manage() or lose_route() before accessing the dialog variables. Cheers, Daniel On 15/01/15 08:25, Will Ferrer wrote: An update on this. I tried setting my dialog module to the use the db. No db entry is ever made. My config looks like this now: loadmodule dialog.so ... modparam(dialog, db_url, DBURL) modparam(dialog, db_mode, 1) modparam(dialog, dlg_flag, 4) modparam(dialog, dlg_match_mode, 1) ... request_route { if (is_method(INVITE) (! has_totag() ) ) { dlg_manage(); } if (is_method(BYE)) { $var(elapsed) = ( $TV(s) - $dlg(start_ts) ); xlog (L_INFO, request_route DIALOG TEST: Completed $dlg(from_uri) to $dlg(to_uri), elapsed: $var(elapsed), now seconds: $TV(s), dlg start time: $dlg(start_ts), DLG_lifetime: $DLG_lifetime); } I hope this message finds every one well. All the best. Will On Thu, Jan 15, 2015 at 12:03 AM, Will Ferrer will.fer...@switchsoft.com wrote: Hi All I am in need of being able to see what the duration of the call was at the time of hang out. I tried turning on the dialog module, but the result is that the values I need are either null or always show as zero. I tried to follow the suggestions in the thread about this here: http://lists.sip-router.org/pipermail/sr-users/2010-October/065889.html In the end my config looks like this: loadmodule dialog.so ... modparam(dialog, dlg_flag, 4) modparam(dialog, dlg_match_mode, 1) ... request_route { if (is_method(INVITE) (! has_totag() ) ) { dlg_manage(); } if (is_method(BYE)) { $var(elapsed) = ( $TV(s) - $dlg(start_ts) ); xlog (L_INFO, request_route DIALOG TEST: Completed $dlg(from_uri) to $dlg(to_uri), elapsed: $var(elapsed), now seconds: $TV(s), dlg start time: $dlg(start_ts), DLG_lifetime: $DLG_lifetime); } Note that I put at the top of the request route just for testing purposes The result I get in
Re: [SR-Users] Dialog module generates error since upgrade to 4.2.2
Klaus, Yes, that solves the problem here as well. Thanks, Jan Klaus Darilion schreef op 2015-01-22 16:16: Hi Jan! I replaced dlg_manage with setflag(dialog flag). This way the dialog is created only when a transaction is created. It solved my problems. regards Klaus On 22.01.2015 14:47, Klaus Darilion wrote: Nice, I have the same problem after upgrading from 4.1.7 to 4.2.2. . in my case it seems the dialog callbacks are not executed for responses ... I am still debugging . On 22.01.2015 13:25, Jan Hazenberg wrote: Hi All, I'm running into a issue with the dialog module since the upgrade from 4.2.1 to 4.2.2. I use the dialog module to add extra vars to the cdr's generated by the ACC module. I use the following config: # - dialog params - modparam(dialog, enable_stats, 1) modparam(dialog, dlg_match_mode, 1) modparam(dialog, dlg_flag, FLT_DLG) modparam(dialog, rr_param, did) modparam(dialog, wait_ack, 1) # Create dialog if (method==INVITE) { # Create Dialog dlg_manage(); # Add test dialog var $dlg_var(src_ua) = $hdr(User-Agent); } This seems to work fine on kamailio 4.2.1 but after the upgrade i see the following errors in the logs: Jan 22 13:16:57 sip /usr/sbin/kamailio[2773]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587@192.168.149.126' and tags '4017642921' '' Jan 22 13:17:01 sip /usr/sbin/kamailio[2770]: CRITICAL: dialog [dlg_hash.c:901]: log_next_state_dlg(): bogus event 7 in state 1 for dlg 0x7f2f0332ed40 [3693:11222] with clid '9192c5fc24627a14c2ec42f084a96587@192.168.149.126' and tags '4017642921' '' If i'm correct bogus event 6 in state 1 indicates that a ACK whas received while the dialog still was in unconfirmed state. Should i catch the 200 OK on the INVITE to update the dialog? Thanks, Jan Hazenberg ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Does forking impact max_inv_lifetime?
I haven't used the feature myself (nor implemented it), but it is what I expect. Maybe you can run in debug mode and map the logs with the code to see what happens. Cheers, Daniel On 22/01/15 00:04, Alex Balashov wrote: Daniel, Will failure_route will be called immediately when the transaction expires? I tried a call where max_inv_lifetime was set at 5 ms and fr_inv_timer at 4 ms. Several new branches were attempted, each sending some 183s or 180s, and Kamailio did not CANCEL the last pending branch until 83 sec into the call. Is there something I'm missing about how to handle the transaction expiration in real time? -- Sent from my BlackBerry. Please excuse errors and brevity. Original Message From: Daniel-Constantin Mierla Sent: Wednesday, January 21, 2015 5:59 PM To: Kamailio (SER) - Users Mailing List Reply To: mico...@gmail.com Subject: Re: [SR-Users] Does forking impact max_inv_lifetime? Hello, somehow your emails are a bit confusing, in first one you say that you cannot get max_inv_lifetime as per transaction, being reset by a new branch, is that still true? The failure route should be called when the transaction is expired. Cheers, Daniel On 21/01/15 18:16, Alex Balashov wrote: Maybe I should ask this question another way that is more applicable to my end-goal: What exactly happens when max_inv_lifetime is reached without a final response? Is a failure_route invoked? If so, is the appropriate means of dealing with this to check t_is_expired() and handle it at that level? -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] CDRs and MongoDB
Hi, Is it possible to write CDRs directly to MongoDB using the new db_mongodb module? modparam(acc, db_url, DB_MONGO_URL) modparam(acc, cdr_enable, 1) modparam(acc, cdr_extra, . ) modparam(acc, cdr_log_enable, 0) modparam(acc, cdrs_table, cdrs) In the acc module documentation, I found the following statement: Note that CDR generation does not involve any kind of database storage (yet). In order to persist the CDRs into a database you will have to set up an exterior process (i.e., a script living outside of Kamailio) and implement the storage task yourself. Thanks, Mickael ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users