Re: [OpenSIPS-Users] CDRTool does not identify by IP origin from OpenSips
Hi Alberto, By default CDRTool takes the gateway parameter out of Source IP (nicely explained also in the doc/RATING.TXT) and if you don't modify the defaults, out of radacct.SourceIP column. In order to identify the gateway, you need to enforce that SourceIP inside opensips script to whatever IP you want to be used as your gateway. Example of params in opensips.cfg: modparam(acc, radius_extra,Source-IP=$avp(s:mygtw)) Another trick you can use is to write your own radius sql query which places the SourceIP from another radius attribute received. Cheers, DanB On Tue, Jul 21, 2009 at 9:44 PM, Alberto Listas lis...@b2br.net wrote: Hi, I am having a difficulty with the rating in CDRTool. When I telnet and give this command: ShowPrice From=sip:005521810...@10.0.0.4 sip%3a005521810...@10.0.0.4 To= sip:00552181000...@10.0.0.1 sip%3a00552181000...@10.0.0.1Gateway=10.0.0.4 Duration=30 (IPs where changed) The rating engine identifies the customer by the gateway and rates correctly. When the call comes from the OpenSips it doesn't identify the customer and uses the default profile: Start time: 2009-07-21 14:55:24 Stop time: 2009-07-21 14:55:59 Method:Invite from : From:5521083200...@10.0.0.4 from%3a5521083200...@10.0.0.4 Domain:10.0.0.4 To (dialed URI):005521810...@10.0.0.4 Canonical URI: 005521810...@10.0.0.1 Next hop URI:005521810...@10.0.0.2 uri%3a005521810...@10.0.0.2 Destination: BRAZIL CELL (55218) Billing Party:5521810...@10.0.0.4 party%3a5521810...@10.0.0.4 Reseller: Duration: 35 s App: audio Destination: 55218 Customer: default Connect: 0. .. Should I set some different variable to FreeRadius for it to identify the GATEWAY? Thanks, Alberto ___ 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] invalid version 0 for table rls_presentity found
I install the opensips-1.5.2-tls when loading module presence it says ERROR:core:db_check_table_version: invalid version 0 for table rls_presentity found, expected 1 and I checked the version table with select * from version, the output is this: | rls_presentity | 0 | | rls_watchers | 2 | how can I change the version of rls_presentity from 0 to 1. I have already migrate opensips_1_4 to opensips_1_5, it's useless -- View this message in context: http://n2.nabble.com/invalid-version-0-for-table-rls_presentity-found-tp3293878p3293878.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] invalid version 0 for table rls_presentity found
2009/7/21 fishy ivy_yuj...@hotmail.com: I install the opensips-1.5.2-tls when loading module presence it says ERROR:core:db_check_table_version: invalid version 0 for table rls_presentity found, expected 1 and I checked the version table with select * from version, the output is this: | rls_presentity | 0 | | rls_watchers | 2 | how can I change the version of rls_presentity from 0 to 1. I have already migrate opensips_1_4 to opensips_1_5, it's useless Most probably OpenSIPS 1.5.2 requires a new version for that table. Create a new database and check it. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] dynamic routing -prefixing on inbound
Hi Julian,
What you need to do is to create some rules to detect some IPs and to
return the prefix to add to RURI. So, see the 1.4.1.2. String
translation (regexp detection, replacement) in
http://www.opensips.org/html/docs/modules/1.5.x/dialplan.html#id227206.
With the prefix returned in a variables (let;s say $var(prefix) ), do :
$rU = $var(prefix) + $rU ;
Regards,
Bogdan
Julien Chavanton wrote:
Do you have a suggestion on how to do this more dynamicaly ?
*From:* Brett Nemeroff [mailto:br...@nemeroff.com]
*Sent:* Tue 21/07/2009 3:12 PM
*To:* Julien Chavanton
*Cc:* Bogdan-Andrei Iancu; users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] dynamic routing -prefixing on inbound
Once again.. another good use for the dialplan module. :)
You really don't want to do this with a static subst I don't think..
-Brett
On Tue, Jul 21, 2009 at 8:30 AM, Julien Chavanton j...@atlastelecom.com
mailto:j...@atlastelecom.com wrote:
Thank you, we will move to 1.6.0 later.
We have partner not sending tech prefix, we need to add it
ourselves, I found this fix, but I wanted to be able to do
everything from OpenSip control panel to keep things manageable.
do_routing(0);
# happend prefix to keep track of originating trunk/gateway
(1.1.1.1==#)
if ( search(From:.*sip:@1.1.1.1 http://1.1.1.1/) ) {
xlog(L_NOTICE, *call
from[1.1.1.1]*\n);
subst_uri('/^sip:([0-9]+)@(.*)$/sip:...@\2;/i');
};
*From:* Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro
mailto:bog...@voice-system.ro]
*Sent:* Tue 21/07/2009 11:06 AM
*To:* Julien Chavanton
*Cc:* users@lists.opensips.org mailto:users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] dynamic routing -prefixing on inbound
Hi Julien,
Yes it is, but only in 1.6.0 (current devel version) . See the
is_from_gw() function:
http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id272676
The 1.5 version has no support for prefixing, but only for stripping:
http://www.opensips.org/html/docs/modules/1.5.x/drouting.html#id272676
Regards,
Bogdan
Julien Chavanton wrote:
I found this in the README of the source code :
* bidirectional behavior - inbound and outbound processing
(strip and prefixing when sending and receiving from a
destination/GW)
I would like to happend a prefix when a call comes in from a certain
gateway, I found that PRI PREFIX is currently applied on when call
it outbound to the gateway but not when a call comes from a gateway.
Is it already doable ?
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Re: [OpenSIPS-Users] rpid_avp and NULL values
Hi Carlo, you must set the line : modparam(auth_db, load_credentials, $avp(s:rpid)=rpid) This line instructs opensips to load at db auth time tht rpid field into the $avp(s:rpid) variable. The line: modparam(auth, rpid_avp, $avp(s:rpid)) simply tells to the append_rpid_hf() function where the RPID value is stored (in what variable). Regards, Bogdan Carlo Dimaggio wrote: Il giorno 20/lug/09, alle ore 19:13, Bogdan-Andrei Iancu ha scritto: Hi Carlo, yes there were some fixes in the 1.5.1 that are now available in 1.5.2. But it is rather strange what you describehow does the load_credentials param look like for you? Hi Bogdan, I have the default value for load_credentials (no param for auth_db) and modparam(auth, rpid_avp, $avp(s:rpid)). Is there any error with this config? Should I need modparam(auth_db, load_credentials, $avp(s:rpid)=rpid) instead of rpid_avp? Thanks and regards, Carlo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] uac_replace_from misses display name
Hi Jeff, you can also do it with uad_replace_from() : http://www.opensips.org/html/docs/modules/1.5.x/uac.html#id228528 See example: # remove display and do not touch uri uac_replace_from(,); Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, I thought had this figured out but it appears not. What's the best way to remove the display name but not touch the rest of the header? Is this another uac_replace_from, a subst operation? - Jeff On 7/9/09 12:04 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, The display part is not restored at all...anyhow it is used only for initial INVITE, so you could simply remove it from all the sequential requests. Regards, Bogdan Jeff Pyle wrote: Hello, I have uac_replace_from updating the display name and URI of outbound calls. The call flow is a standard INVITE, 100, 183, 200, ACK kind of setup. The module seems to be correctly replacing the names up to the ACK. When it forwards the ACK from the UAS back to the UAC, it updates the URI but not the display name. The BYE and 200 OK to tear down the call also have this problem. This is on opensips 1.5.1-notls (sparc64/linux). Any thoughts? Thanks, Jeff ___ 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] invalid version 0 for table rls_presentity found
Hi, are you installing 1.5.2 or upgrading an existing 1.4 ? Because, as Inaki said, you current DB is not compatible with the version of opensips you are running. If installing, create from scratch the db via opensipsdbctl create command. If upgrading from 1.4, do opensipsdbctl migrate. Regrads, Bogdan fishy wrote: I install the opensips-1.5.2-tls when loading module presence it says ERROR:core:db_check_table_version: invalid version 0 for table rls_presentity found, expected 1 and I checked the version table with select * from version, the output is this: | rls_presentity | 0 | | rls_watchers | 2 | how can I change the version of rls_presentity from 0 to 1. I have already migrate opensips_1_4 to opensips_1_5, it's useless ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to add an IMPU as subscriber with username (sip:1...@example.com)?
Hi Cao, The correct format for adding a user is an SIP AOR and not a SIP URI. Do: opensipsctl add 1...@example.com 108108 Regards, Bogdan Cao Lei-MNW784 wrote: Hi, Everyone! I tried to add a new user using: opensipsctl add sip:1...@example.com 108108 However, in the database, the new entry is added as: | id | username | domain | password | email_address | | 42 | sip:108 | example.com | 108108 | | I want to use the full IMPU format sip:1...@example.com to do digest authentication. But the username configured in database is sip:108. Any suggestion that how to add a subscriber so that it has username sip:1...@example.com? Thanks Regards Cao, Charles ___ 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] Storing modules parameters on AVP, as much as I can
Hi Ricardo, there are things here: 1) how to set the value of the parameters (via DB, instead of statically hardcoding in cfg) 2) dynamic changing of the module params without restart. What case are you refering at? Regards, Bogdan Ricardo Martins wrote: Hi all! Do anybody knows if there is a trick to store the modules parameters on database/avps? I want to give the opensips administrator all the flexibility I can without having to edit cfg text file. I know that there is some specific parameters where you can do that like fr_inv_timer or fr_timer for tm module but I'm talking about storing other parameters that don't have an specific avp enable mode like T1 and T2, for example. Do anybody has any clue? Regards! Ricardo. ___ 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] uac_replace_from misses display name
Indeed. I missed that example... Thanks. On 7/22/09 5:11 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, you can also do it with uad_replace_from() : http://www.opensips.org/html/docs/modules/1.5.x/uac.html#id228528 See example: # remove display and do not touch uri uac_replace_from(,); Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] codec manipulation feature
Hi Bogdan, Just curious if you had a chance to see what the status is of this SDP manipulation code. Thanks, Jeff On 6/15/09 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, I have some code from last year - let me check it out and see what is the status. Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It¹s been a little while since we talked about this. I was wondering if there was anything in the works to detect and/or manipulate the codecs present in an SDP. - Jeff On 2/1/09 4:08 AM, Steve Kurzeja steve.kurz...@gmail.com wrote: This idea is quite standard in SBCs, typically called codec profiles, where you permit only certain codecs to be passed through the SBC in an INVITE and all the rest are stripped out. We use it to get around interop issues with certain codecs. E.g. we have some end devices/customers that have issues using g729a so we choose to remove this codec for these specific endpoints. The poor man's method to implementing this is just doing header manipulations in the SDP but it would be nice to be standardized. Regards, Steve On Fri, Jan 30, 2009 at 2:20 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hi Bogdan, I'm looking for the ability to selectively remove codec advertisements from the SDP. For example, if my customer sends a call to me for PSTN termination he may advertise G711 and G729, with G711 preferred. By looking at the number of existing dialogs I may know that he's running low on bandwidth, so I would like to suppress the G711 advertisement ultimately causing a 200 OK from the carrier with G729. Generically, in this application we're looking only to suppress G711 at certain times. I understand normally codec selection is done completely by the gateway device. However, my gateway devices aren't smart enough to take bandwidth utilization into consideration when choosing which codecs to advertise. I'm hoping my proxy might be. :) Does that make sense? - Jeff On 1/29/09 5:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, right now there is only available some functionality to check the codecs (to see what codecs are advertised in the SDP)... What exactly are you looking for (like codec ops) ? Regards, Bogdan Jeff Pyle wrote: Bogdan, Some months back you mentioned an upcoming feature that would allow Opensips to manipulate the codecs present in the SDP. Just wondering if there is anything available to test yet. This feature, in combination with dialog contexts, will be of great use to us to allow us to take a guess at the bandwidth consumption for a particular customer and force the use of a compressed codec if necessary. Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] codec manipulation feature
Hi Jeff, This is Andrei. I am currently working on this right now. It should be finished in a couple of days if nothing comes up. Andrei. --- On Wed, 7/22/09, Jeff Pyle jp...@fidelityvoice.com wrote: Just curious if you had a chance to see what the status is of this SDP manipulation code. Thanks, Jeff On 6/15/09 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, I have some code from last year - let me check it out and see what is the status. Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It¹s been a little while since we talked about this. I was wondering if there was anything in the works to detect and/or manipulate the codecs present in an SDP. - Jeff On 2/1/09 4:08 AM, Steve Kurzeja steve.kurz...@gmail.com wrote: This idea is quite standard in SBCs, typically called codec profiles, where you permit only certain codecs to be passed through the SBC in an INVITE and all the rest are stripped out. We use it to get around interop issues with certain codecs. E.g. we have some end devices/customers that have issues using g729a so we choose to remove this codec for these specific endpoints. The poor man's method to implementing this is just doing header manipulations in the SDP but it would be nice to be standardized. Regards, Steve On Fri, Jan 30, 2009 at 2:20 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hi Bogdan, I'm looking for the ability to selectively remove codec advertisements from the SDP. For example, if my customer sends a call to me for PSTN termination he may advertise G711 and G729, with G711 preferred. By looking at the number of existing dialogs I may know that he's running low on bandwidth, so I would like to suppress the G711 advertisement ultimately causing a 200 OK from the carrier with G729. Generically, in this application we're looking only to suppress G711 at certain times. I understand normally codec selection is done completely by the gateway device. However, my gateway devices aren't smart enough to take bandwidth utilization into consideration when choosing which codecs to advertise. I'm hoping my proxy might be. :) Does that make sense? - Jeff On 1/29/09 5:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, right now there is only available some functionality to check the codecs (to see what codecs are advertised in the SDP)... What exactly are you looking for (like codec ops) ? Regards, Bogdan Jeff Pyle wrote: Bogdan, Some months back you mentioned an upcoming feature that would allow Opensips to manipulate the codecs present in the SDP. Just wondering if there is anything available to test yet. This feature, in combination with dialog contexts, will be of great use to us to allow us to take a guess at the bandwidth consumption for a particular customer and force the use of a compressed codec if necessary. Thanks, Jeff ___ 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] codec manipulation feature
Great, Andrei. I look forward to being able to test it. - Jeff On 7/22/09 8:59 AM, andrei dragus andreidra...@yahoo.com wrote: Hi Jeff, This is Andrei. I am currently working on this right now. It should be finished in a couple of days if nothing comes up. Andrei. --- On Wed, 7/22/09, Jeff Pyle jp...@fidelityvoice.com wrote: Just curious if you had a chance to see what the status is of this SDP manipulation code. Thanks, Jeff On 6/15/09 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, I have some code from last year - let me check it out and see what is the status. Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It¹s been a little while since we talked about this. I was wondering if there was anything in the works to detect and/or manipulate the codecs present in an SDP. - Jeff On 2/1/09 4:08 AM, Steve Kurzeja steve.kurz...@gmail.com wrote: This idea is quite standard in SBCs, typically called codec profiles, where you permit only certain codecs to be passed through the SBC in an INVITE and all the rest are stripped out. We use it to get around interop issues with certain codecs. E.g. we have some end devices/customers that have issues using g729a so we choose to remove this codec for these specific endpoints. The poor man's method to implementing this is just doing header manipulations in the SDP but it would be nice to be standardized. Regards, Steve On Fri, Jan 30, 2009 at 2:20 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hi Bogdan, I'm looking for the ability to selectively remove codec advertisements from the SDP. For example, if my customer sends a call to me for PSTN termination he may advertise G711 and G729, with G711 preferred. By looking at the number of existing dialogs I may know that he's running low on bandwidth, so I would like to suppress the G711 advertisement ultimately causing a 200 OK from the carrier with G729. Generically, in this application we're looking only to suppress G711 at certain times. I understand normally codec selection is done completely by the gateway device. However, my gateway devices aren't smart enough to take bandwidth utilization into consideration when choosing which codecs to advertise. I'm hoping my proxy might be. :) Does that make sense? - Jeff On 1/29/09 5:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, right now there is only available some functionality to check the codecs (to see what codecs are advertised in the SDP)... What exactly are you looking for (like codec ops) ? Regards, Bogdan Jeff Pyle wrote: Bogdan, Some months back you mentioned an upcoming feature that would allow Opensips to manipulate the codecs present in the SDP. Just wondering if there is anything available to test yet. This feature, in combination with dialog contexts, will be of great use to us to allow us to take a guess at the bandwidth consumption for a particular customer and force the use of a compressed codec if necessary. Thanks, Jeff ___ 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] codec manipulation feature
Hello Andrei, Are you using the core sdp parser implementation? Ovidiu On Wed, Jul 22, 2009 at 8:59 AM, andrei dragusandreidra...@yahoo.com wrote: Hi Jeff, This is Andrei. I am currently working on this right now. It should be finished in a couple of days if nothing comes up. Andrei. --- On Wed, 7/22/09, Jeff Pyle jp...@fidelityvoice.com wrote: Just curious if you had a chance to see what the status is of this SDP manipulation code. Thanks, Jeff On 6/15/09 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, I have some code from last year - let me check it out and see what is the status. Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It¹s been a little while since we talked about this. I was wondering if there was anything in the works to detect and/or manipulate the codecs present in an SDP. - Jeff On 2/1/09 4:08 AM, Steve Kurzeja steve.kurz...@gmail.com wrote: This idea is quite standard in SBCs, typically called codec profiles, where you permit only certain codecs to be passed through the SBC in an INVITE and all the rest are stripped out. We use it to get around interop issues with certain codecs. E.g. we have some end devices/customers that have issues using g729a so we choose to remove this codec for these specific endpoints. The poor man's method to implementing this is just doing header manipulations in the SDP but it would be nice to be standardized. Regards, Steve On Fri, Jan 30, 2009 at 2:20 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hi Bogdan, I'm looking for the ability to selectively remove codec advertisements from the SDP. For example, if my customer sends a call to me for PSTN termination he may advertise G711 and G729, with G711 preferred. By looking at the number of existing dialogs I may know that he's running low on bandwidth, so I would like to suppress the G711 advertisement ultimately causing a 200 OK from the carrier with G729. Generically, in this application we're looking only to suppress G711 at certain times. I understand normally codec selection is done completely by the gateway device. However, my gateway devices aren't smart enough to take bandwidth utilization into consideration when choosing which codecs to advertise. I'm hoping my proxy might be. :) Does that make sense? - Jeff On 1/29/09 5:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, right now there is only available some functionality to check the codecs (to see what codecs are advertised in the SDP)... What exactly are you looking for (like codec ops) ? Regards, Bogdan Jeff Pyle wrote: Bogdan, Some months back you mentioned an upcoming feature that would allow Opensips to manipulate the codecs present in the SDP. Just wondering if there is anything available to test yet. This feature, in combination with dialog contexts, will be of great use to us to allow us to take a guess at the bandwidth consumption for a particular customer and force the use of a compressed codec if necessary. Thanks, Jeff ___ 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] codec manipulation feature
Yes. --- On Wed, 7/22/09, Ovidiu Sas o...@voipembedded.com wrote: From: Ovidiu Sas o...@voipembedded.com Hello Andrei, Are you using the core sdp parser implementation? Ovidiu On Wed, Jul 22, 2009 at 8:59 AM, andrei dragusandreidra...@yahoo.com wrote: Hi Jeff, This is Andrei. I am currently working on this right now. It should be finished in a couple of days if nothing comes up. Andrei. --- On Wed, 7/22/09, Jeff Pyle jp...@fidelityvoice.com wrote: Just curious if you had a chance to see what the status is of this SDP manipulation code. Thanks, Jeff On 6/15/09 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, I have some code from last year - let me check it out and see what is the status. Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It¹s been a little while since we talked about this. I was wondering if there was anything in the works to detect and/or manipulate the codecs present in an SDP. - Jeff On 2/1/09 4:08 AM, Steve Kurzeja steve.kurz...@gmail.com wrote: This idea is quite standard in SBCs, typically called codec profiles, where you permit only certain codecs to be passed through the SBC in an INVITE and all the rest are stripped out. We use it to get around interop issues with certain codecs. E.g. we have some end devices/customers that have issues using g729a so we choose to remove this codec for these specific endpoints. The poor man's method to implementing this is just doing header manipulations in the SDP but it would be nice to be standardized. Regards, Steve On Fri, Jan 30, 2009 at 2:20 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hi Bogdan, I'm looking for the ability to selectively remove codec advertisements from the SDP. For example, if my customer sends a call to me for PSTN termination he may advertise G711 and G729, with G711 preferred. By looking at the number of existing dialogs I may know that he's running low on bandwidth, so I would like to suppress the G711 advertisement ultimately causing a 200 OK from the carrier with G729. Generically, in this application we're looking only to suppress G711 at certain times. I understand normally codec selection is done completely by the gateway device. However, my gateway devices aren't smart enough to take bandwidth utilization into consideration when choosing which codecs to advertise. I'm hoping my proxy might be. :) Does that make sense? - Jeff On 1/29/09 5:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, right now there is only available some functionality to check the codecs (to see what codecs are advertised in the SDP)... What exactly are you looking for (like codec ops) ? Regards, Bogdan Jeff Pyle wrote: Bogdan, Some months back you mentioned an upcoming feature that would allow Opensips to manipulate the codecs present in the SDP. Just wondering if there is anything available to test yet. This feature, in combination with dialog contexts, will be of great use to us to allow us to take a guess at the bandwidth consumption for a particular customer and force the use of a compressed codec if necessary. Thanks, Jeff ___ 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] codec manipulation feature
Perfect. Let me know if you need more features from the sdp parser. Thanks, Ovidiu On Wed, Jul 22, 2009 at 11:32 AM, andrei dragusandreidra...@yahoo.com wrote: Yes. --- On Wed, 7/22/09, Ovidiu Sas o...@voipembedded.com wrote: From: Ovidiu Sas o...@voipembedded.com Hello Andrei, Are you using the core sdp parser implementation? Ovidiu On Wed, Jul 22, 2009 at 8:59 AM, andrei dragusandreidra...@yahoo.com wrote: Hi Jeff, This is Andrei. I am currently working on this right now. It should be finished in a couple of days if nothing comes up. Andrei. --- On Wed, 7/22/09, Jeff Pyle jp...@fidelityvoice.com wrote: Just curious if you had a chance to see what the status is of this SDP manipulation code. Thanks, Jeff On 6/15/09 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, I have some code from last year - let me check it out and see what is the status. Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It¹s been a little while since we talked about this. I was wondering if there was anything in the works to detect and/or manipulate the codecs present in an SDP. - Jeff On 2/1/09 4:08 AM, Steve Kurzeja steve.kurz...@gmail.com wrote: This idea is quite standard in SBCs, typically called codec profiles, where you permit only certain codecs to be passed through the SBC in an INVITE and all the rest are stripped out. We use it to get around interop issues with certain codecs. E.g. we have some end devices/customers that have issues using g729a so we choose to remove this codec for these specific endpoints. The poor man's method to implementing this is just doing header manipulations in the SDP but it would be nice to be standardized. Regards, Steve On Fri, Jan 30, 2009 at 2:20 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hi Bogdan, I'm looking for the ability to selectively remove codec advertisements from the SDP. For example, if my customer sends a call to me for PSTN termination he may advertise G711 and G729, with G711 preferred. By looking at the number of existing dialogs I may know that he's running low on bandwidth, so I would like to suppress the G711 advertisement ultimately causing a 200 OK from the carrier with G729. Generically, in this application we're looking only to suppress G711 at certain times. I understand normally codec selection is done completely by the gateway device. However, my gateway devices aren't smart enough to take bandwidth utilization into consideration when choosing which codecs to advertise. I'm hoping my proxy might be. :) Does that make sense? - Jeff On 1/29/09 5:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, right now there is only available some functionality to check the codecs (to see what codecs are advertised in the SDP)... What exactly are you looking for (like codec ops) ? Regards, Bogdan Jeff Pyle wrote: Bogdan, Some months back you mentioned an upcoming feature that would allow Opensips to manipulate the codecs present in the SDP. Just wondering if there is anything available to test yet. This feature, in combination with dialog contexts, will be of great use to us to allow us to take a guess at the bandwidth consumption for a particular customer and force the use of a compressed codec if necessary. Thanks, Jeff ___ 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] codec manipulation feature
Hi Jeff, Andrei, the author of the memchached module is working on this right now. I guess this will be ready soon ;) Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, Just curious if you had a chance to see what the status is of this SDP manipulation code. Thanks, Jeff On 6/15/09 6:36 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, I have some code from last year - let me check it out and see what is the status. Regards, Bogdan Jeff Pyle wrote: Hi Bogdan, It¹s been a little while since we talked about this. I was wondering if there was anything in the works to detect and/or manipulate the codecs present in an SDP. - Jeff On 2/1/09 4:08 AM, Steve Kurzeja steve.kurz...@gmail.com wrote: This idea is quite standard in SBCs, typically called codec profiles, where you permit only certain codecs to be passed through the SBC in an INVITE and all the rest are stripped out. We use it to get around interop issues with certain codecs. E.g. we have some end devices/customers that have issues using g729a so we choose to remove this codec for these specific endpoints. The poor man's method to implementing this is just doing header manipulations in the SDP but it would be nice to be standardized. Regards, Steve On Fri, Jan 30, 2009 at 2:20 AM, Jeff Pyle jp...@fidelityvoice.com wrote: Hi Bogdan, I'm looking for the ability to selectively remove codec advertisements from the SDP. For example, if my customer sends a call to me for PSTN termination he may advertise G711 and G729, with G711 preferred. By looking at the number of existing dialogs I may know that he's running low on bandwidth, so I would like to suppress the G711 advertisement ultimately causing a 200 OK from the carrier with G729. Generically, in this application we're looking only to suppress G711 at certain times. I understand normally codec selection is done completely by the gateway device. However, my gateway devices aren't smart enough to take bandwidth utilization into consideration when choosing which codecs to advertise. I'm hoping my proxy might be. :) Does that make sense? - Jeff On 1/29/09 5:04 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, right now there is only available some functionality to check the codecs (to see what codecs are advertised in the SDP)... What exactly are you looking for (like codec ops) ? Regards, Bogdan Jeff Pyle wrote: Bogdan, Some months back you mentioned an upcoming feature that would allow Opensips to manipulate the codecs present in the SDP. Just wondering if there is anything available to test yet. This feature, in combination with dialog contexts, will be of great use to us to allow us to take a guess at the bandwidth consumption for a particular customer and force the use of a compressed codec if necessary. Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Storing modules parameters on AVP, as much as I can
Hi Bogdan. The first option would be enough. I don't think that's an issue a restarting procedure just after changing parameters. DanB just wrote me indicating an script generator solution. Do you think that would be possible to use another solution? About the second option, I know that I can do it with fr_timer_avp or fr_inv_timer_avp, right? Regards, Ricardo. Bogdan-Andrei Iancu escreveu: Hi Ricardo, there are things here: 1) how to set the value of the parameters (via DB, instead of statically hardcoding in cfg) 2) dynamic changing of the module params without restart. What case are you refering at? Regards, Bogdan Ricardo Martins wrote: Hi all! Do anybody knows if there is a trick to store the modules parameters on database/avps? I want to give the opensips administrator all the flexibility I can without having to edit cfg text file. I know that there is some specific parameters where you can do that like fr_inv_timer or fr_timer for tm module but I'm talking about storing other parameters that don't have an specific avp enable mode like T1 and T2, for example. Do anybody has any clue? Regards! Ricardo. ___ 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] CDRTool does not identify by IP origin from OpenSips
Hi Dan, Thanks for the info, the doc/Rating.txt is very good but sometimes you need previous experience to understand the details. I was already using the param: Source-IP=$avp(s:source_ip); But I wasn't setting any values so the SourceIP column was blank. I added a line to the cfg to set the value: $avp(s:source_ip) = $si; And now the rating identifies the origin by SourceIP when the CallingStationId is in the form 123...@1.1.1.1 (and hence the domain is set to the IP of the source) but when CallingStationId is in the form 1.1.1.1 (just the IP) the Domain in the info of the call is blank and the rating does not work. When the Domain is blank the rating engine use the profile Customer: al...@example.com from the default Customer configuration of CDRTool. I suppose this is because this customer has a blank domain field. But this is an error since it should use the default profile instead. Anyway, Thanks for your help, now it can rate by IP. Cheers, Alberto - Original Message - From: DanB To: Alberto Listas Cc: users@lists.opensips.org Sent: Wednesday, July 22, 2009 4:23 AM Subject: Re: [OpenSIPS-Users] CDRTool does not identify by IP origin from OpenSips Hi Alberto, By default CDRTool takes the gateway parameter out of Source IP (nicely explained also in the doc/RATING.TXT) and if you don't modify the defaults, out of radacct.SourceIP column. In order to identify the gateway, you need to enforce that SourceIP inside opensips script to whatever IP you want to be used as your gateway. Example of params in opensips.cfg: modparam(acc, radius_extra,Source-IP=$avp(s:mygtw)) Another trick you can use is to write your own radius sql query which places the SourceIP from another radius attribute received. Cheers, DanB On Tue, Jul 21, 2009 at 9:44 PM, Alberto Listas lis...@b2br.net wrote: Hi, I am having a difficulty with the rating in CDRTool. When I telnet and give this command: ShowPrice From=sip:005521810...@10.0.0.4 To=sip:00552181000...@10.0.0.1 Gateway=10.0.0.4 Duration=30 (IPs where changed) The rating engine identifies the customer by the gateway and rates correctly. When the call comes from the OpenSips it doesn't identify the customer and uses the default profile: Start time: 2009-07-21 14:55:24 Stop time: 2009-07-21 14:55:59 Method:Invite from : From:5521083200...@10.0.0.4 Domain:10.0.0.4 To (dialed URI):005521810...@10.0.0.4 Canonical URI: 005521810...@10.0.0.1 Next hop URI:005521810...@10.0.0.2 Destination: BRAZIL CELL (55218) Billing Party:5521810...@10.0.0.4 Reseller: Duration: 35 s App: audio Destination: 55218 Customer: default Connect: 0. .. Should I set some different variable to FreeRadius for it to identify the GATEWAY? Thanks, Alberto ___ 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] uac_replace_from() mangling the From header
Hello, In previous installations I've stayed away from uac_replace_from() because in many cases I've seen it mangle the From field to the point where it's out of spec. This time I have to use it, and I'm not having much success. In addition to loading the appropriate modules, I have configured: modparam(rr, append_fromtag, 1) modparam(uac,from_passwd,my_password) modparam(uac,from_restore_mode,auto) I've read running it more than once on a transaction is bad news. I'm not doing that. I'm running it once, something like this: $var(fromuri) = sip:anonymous@ + $Ri; uac_replace_from(anonymous, $var(fromuri)); I place an outbound call where this logic is hit, and all is well. It's the restore that happens on a loose-routed BYE from the network side when the far end hangs up that seems to get things a bit out of whack. The To field in this case, since it's backwards, is being restored as follows, according to ngrep: To: Voice Lab sip:9998880...@11.22.33.44.:5060;transport=UDP;tag=2378b50-0-13c4-6a31 What's interesting is the 11.22.33.44 address belongs to the upstream proxy that handed the call termination, not the originating one. My CPE equipment is complaining of an extra NULL character at the end of the address, which ngrep shows above as a .. I've tried it with variables, with static values (sip:anonym...@anonymous.invalid)... Something always gets mangled in a most unfortunate way. This is on Opensips 1.5.1 on sparc. Any suggestions would be most appreciated. Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
