Re: [OpenSIPS-Users] Query regarding AWS document BD using through Opensips .

[Kevin Wormington]

Hi Samita,

I don’t have any experience with AWS but from the error message OpenSIPS is 
logging the hostname of the server cannot be resolved.  The hostname also 
appears to be truncated.   Have you tried using the IP address instead of 
hostname or making a CNAME dns entry for the host that is shorter?   Perhaps 
this is some parameter length limit.

Kevin
> On Apr 29, 2022, at 8:23 AM, Sasmita Panda  wrote:
> 
> Hi All , 
> 
> 
> I was exploring fullsharing-cachedb-cluster in opensips 3.2 . I have tested 
> this with single stand alone mongo db instance . Its working perfectly fine . 
> 
> I know that AWS document DB is mongodb compatible . So I want to explore that 
> . Because we are using AWS cloud for our deployment . 
> 
> I have created a single instance of Document DB cluster . I want to connect 
> to that from the opensips script . There was no error while starting opensips 
> . But when opensips tried to write data in the db its threw an error . 
> 
> 
>  ERROR:usrloc:release_urecord: failed to flush AoR 
> default_line_11...@p2p-cachedb.xyz.com
>  ERROR:cachedb_mongodb:mongo_con_update: last error: 15.13053: No suitable 
> servers found (`serverselectiontryonce` set): [Failed to resolve 
> 'docdb-2022-04-27-10-26-28.cluster-cryhhicuxgzu.us-east-1.docdb.amdocdb-2022-04-27-10-26-28.cluster-cryhhicuxgzu.us-east-1.docdb.amazona']
>  ERROR:usrloc:cdb_flush_urecord: cache update query for AoR 
> hynode2_calldefa...@p2p-cachedb.xyz.com failed!
> 
> 
> My configuration file looks like below  . same for usrloc and db_cachedb
> 
> modparam("cachedb_mongodb", "cachedb_url","mongodb://
> ?master:opensi...@docdb-2022.cluster-cryhhicuxgzu.us-east-1.docdb.amazonaws.com:27017/db.test/
> ?ssl=true&ssl_ca_certs=/usr/local/src/etc/opensips/rds-combined-ca-bundle.pem&replicaSet=rs0&readPreference=secondaryPreferred&retryWrites=false")
> 
> In document db it also its says the same way we can connect to docdb from an 
> application . 
> 
> I have tried so many ways to resolve this . But without any luck . Please 
> help me out if anybody has used a document db through any application can 
> also reply . 
> 
> 
> Thanks & Regards
> Sasmita Panda
> Senior Network Testing and Software Engineer
> 3CLogic , ph:07827611765
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Re: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6

[Kevin Wormington]

AVP is for sure supported on 3.1.x and up.  We use the following:

uac_replace_from($avp(caller_cnam),"”);

> On Mar 8, 2022, at 8:12 AM, Alain Bieuzent  wrote:
> 
> Hmm , not sure $avp is supported can you try with $var
>  
> $var(ds)="abc";
> uac_replace_from($var(ds),""); 
> 
>  
> De : Users  au nom de Vinayak Makwana 
> 
> Répondre à : OpenSIPS users mailling list 
> Date : mardi 8 mars 2022 à 14:01
> À : 
> Objet : Re: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6
>  
> Hello Alain Bieuzent
> 
> I tried with the uac_replace_from() function also but not getting the proper 
> result.
> 
> Here's my input & output result:
> INPUT:  From:"abc";tag=6a8eda3f
> OUTPUT:  From:abc;tag=6a8eda3f
> 
> Here's my logic: 
> $avp(ds)="abc";
> uac_replace_from($avp(ds),""); 
> 
> So Can you please tell me what is an issue why not getting quotes
> 
> Thanks in advance 
> Vinayak Makwana
>  
> On Tue, Mar 8, 2022 at 5:30 PM  wrote:
>> Send Users mailing list submissions to
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>> 
>> Today's Topics:
>> 
>>1. Header manipulation (Vinayak Makwana)
>>2. Re: Header manipulation (Alain Bieuzent)
>> 
>> 
>> --
>> 
>> Message: 1
>> Date: Tue, 8 Mar 2022 16:11:12 +0530
>> From: Vinayak Makwana 
>> To: users@lists.opensips.org
>> Subject: [OpenSIPS-Users] Header manipulation
>> Message-ID:
>> 
>> Content-Type: text/plain; charset="utf-8"
>> 
>> Hello All,
>> 
>> I want to replace uri-display in the FROM header using the avp_subst
>> function. So, can anyone suggest a solution ?
>> 
>> Here's My scenario:
>> Main:->  From:"abc";tag=6a8eda3f
>> After Changes ->  From:"pqrs"> ;transport=UDP>;tag=6a8eda3f
>> 
>> Many Thanks
>> Vinayak Makwana
>> 
>> -- 
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>> Message: 2
>> Date: Tue, 08 Mar 2022 11:51:45 +0100
>> From: Alain Bieuzent 
>> To: OpenSIPS users mailling list 
>> Subject: Re: [OpenSIPS-Users] Header manipulation
>> Message-ID: 
>> Content-Type: text/plain; charset="utf-8"
>> 
>> Hi Vinayak,
>> 
>> 
>> 
>> Try with uac_replace_from([display],uri)
>> 
>> 
>> 
>> Ragards
>> 
>> 
>> 
>> De : Users  au nom de Vinayak Makwana 
>> 
>> Répondre à : OpenSIPS users mailling list 
>> Date : mardi 8 mars 2022 à 11:43
>> À : 
>> Objet : [OpenSIPS-Users] Header manipulation
>> 
>> 
>> 
>> Hello All,
>> 
>> I want to replace uri-display in the FROM header using the avp_subst 
>> function. So, can anyone suggest a solution ?
>> 
>> Here's My scenario:
>> Main:->  From:"abc";tag=6a8eda3f 
>> After Changes ->  
>> From:"pqrs";tag=6a8eda3f
>> 
>> Many Thanks
>> Vinayak Makwana
>> 
>> 
>> 
>> Disclaimer
>> 
>> In addition to generic Disclaimer which you have agreed on our website, any 
>> views or opinions presented in this email are solely those of the originator 
>> an

Re: [OpenSIPS-Users] opensips_port_change

[Kevin Wormington]

Hi Murat,

Assuming you are using 3.x.x version you would set it in opensips.cfg as 
follows:

socket=udp:192.168.0.1:5080
socket=tcp:192.168.0.1:5080

Obviously you would replace the 192.168.0.1 with your IP and the 5080 with your 
desired port.

Hope this helps,

Kevin
> On Jun 2, 2021, at 8:43 AM, murat bal  wrote:
> 
> hello, can the port of opensips server 5060 be changed? I tried with 
> opensips.cfg, it didn't work. Can you help me? Thanks. 
> 
> 
> 
> Murat
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Re: [OpenSIPS-Users] Opensips CP Permissions "RELOAD on SERVER" produces error

[Kevin Wormington]

I would start with double checking that the permissions module is actually 
loaded on the server.  That is the most common reason to get a reload error.

Kevin
> On Jul 20, 2021, at 11:26 AM, Jeff Wilkie  wrote:
> 
> Opensips 3.1.2
> CP 8.3.1
> Debian 10
> 
> When adding permissions, I hit "Reload on Server" but I get the following 
> error:
> 
> Sending to json:127.0.0.1:/mi : MI command failed with code -32601 
> (Method not found)
> 
> I​
> I don't get this error on any other page but this one when attempting to 
> "Reload on Server".  Is there something specifically wrong with this page and 
> how it uses the MI command structure?  Again, All other pages that use the 
> "Reload on Server" give a 200ok and work as expected.
> 
> Thanks,
> 
> Jeff
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Re: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on node failure

[Kevin Wormington]

I’m not using RTPEngine…the upstream proxies are handling all media, NAT 
traversal, etc. so the OpenSIPS instances can always reach the endpoints.  I’m 
using clusterer module to share the user location and dialogs with different 
active tags per node.  There is zero loss of media on switch-over and sometimes 
a little longer PDD for new calls during switchover until the upstream proxies 
detect the instance down.   The only part I can’t seem to get to work is 
handling the final BYE for calls that were on the failed node originally.   The 
re-invite ping will correct end them but would like to be able to fix it 
completely…but maybe that is not currently possible.


Thanks,

Kevin
> On Jan 19, 2021, at 9:31 AM, Social Boh via Users  
> wrote:
> 
> To switch calls from one server to another you have to use redis and 
> rptengine using HA with pacemaker y corosync.
> 
> You must have two OpenSIPs, Two RTPEngine, Two Redis servers 
> (primary-replica) Two Mariad  servers (primary/primary)
> 
> With redis you can save calls data (ip, ports, callid) on active server and 
> then use these data on the replica server when swithc to active. On my tests, 
> when switching from a server to another I have between 5 and 10 seconds 
> without audio.
> 
> Regards
> 
> ---
> I'm SoCIaL, MayBe
> 
> El 19/01/2021 a las 10:00 a. m., Kevin Wormington escribió:
>> I’m not using a VIP and I have made some progress by setting a different 
>> active tag on each node…then upon node failure setting the failed node's tag 
>> to active on remaining node.  This lets the re-invite pinging work, etc.  
>> It’s almost there but the handling of the BYE…they are still sent to the IP 
>> of the failed node even after re-invite pings so any in-progress calls from 
>> the failed node are zombie when they hang up until the re-invite ping times 
>> out (30 seconds).   I found an article about initiating a re-invite on the 
>> new node with something like "opensips-cli -x mi dlg_send_sequential 
>> callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” 
>> but that either seems to terminate the call immediately or say the dialog 
>> wasn’t found.
>> 
>> 
>> Thanks,
>> 
>> Kevin
>>> On Jan 19, 2021, at 8:46 AM, Andy Dierlam  wrote:
>>> 
>>> With dialog writing to db that both servers use.   And same tag on both - 
>>> modparam("dialog", "dlg_sharing_tag", "vip1=active")
>>> had this working on opensips 2.4
>>> 
>>> thanks
>>> Andy
>>> 
>>> 
>>> On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington 
>>>  wrote:
>>> Hi,
>>> 
>>> I've been attempting to get a two node active/active setup to work with the 
>>> v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a 
>>> proxy that handles all of the NAT/media so either OpenSIPS instance can 
>>> communicate directly with the user.
>>> 
>>> What I have working so far:
>>> 
>>> Registrations and calls work when sent to either node and if you stop 
>>> OpenSIPS on a node new calls work fine using the other node.
>>> 
>>> What I can’t get to work:
>>> 
>>> Calls that are already in progress to switch between nodes when one node 
>>> fails.
>>> 
>>> 
>>> I have messed around with various sharing tags…no tag, same tag, different 
>>> tags but haven’t had any luck.   I’m guessing that I’m missing something to 
>>> trigger the remaining node to send re-invites.  Has anyone attempted this 
>>> type of setup and have any ideas?
>>> 
>>> Thanks,
>>> 
>>> Kevin
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Re: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on node failure

[Kevin Wormington]

That seems to be how most are setup…maybe I’m making it harder than it should 
be :-)

Out of curiosity what did/do you use to monitor OpenSIPS as up for your 
failover or did you just rely on the IP (keepalived, etc.) reachability? 

Thanks,

Kevin
> On Jan 19, 2021, at 9:08 AM, Andy Dierlam  wrote:
> 
> Ah, my setup was with a floating IP between servers. 
> 
> thanks
> Andy
> 
> On Tue, Jan 19, 2021 at 10:02 AM Kevin Wormington 
>  wrote:
> I’m not using a VIP and I have made some progress by setting a different 
> active tag on each node…then upon node failure setting the failed node's tag 
> to active on remaining node.  This lets the re-invite pinging work, etc.  
> It’s almost there but the handling of the BYE…they are still sent to the IP 
> of the failed node even after re-invite pings so any in-progress calls from 
> the failed node are zombie when they hang up until the re-invite ping times 
> out (30 seconds).   I found an article about initiating a re-invite on the 
> new node with something like "opensips-cli -x mi dlg_send_sequential 
> callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” but 
> that either seems to terminate the call immediately or say the dialog wasn’t 
> found.
> 
> 
> Thanks,
> 
> Kevin
> > On Jan 19, 2021, at 8:46 AM, Andy Dierlam  wrote:
> > 
> > With dialog writing to db that both servers use.   And same tag on both - 
> > modparam("dialog", "dlg_sharing_tag", "vip1=active")
> > had this working on opensips 2.4 
> > 
> > thanks
> > Andy
> > 
> > 
> > On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington 
> >  wrote:
> > Hi,
> > 
> > I've been attempting to get a two node active/active setup to work with the 
> > v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a 
> > proxy that handles all of the NAT/media so either OpenSIPS instance can 
> > communicate directly with the user.  
> > 
> > What I have working so far:
> > 
> > Registrations and calls work when sent to either node and if you stop 
> > OpenSIPS on a node new calls work fine using the other node.
> > 
> > What I can’t get to work:
> > 
> > Calls that are already in progress to switch between nodes when one node 
> > fails.
> > 
> > 
> > I have messed around with various sharing tags…no tag, same tag, different 
> > tags but haven’t had any luck.   I’m guessing that I’m missing something to 
> > trigger the remaining node to send re-invites.  Has anyone attempted this 
> > type of setup and have any ideas?
> > 
> > Thanks,
> > 
> > Kevin
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> 
> 
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Re: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on node failure

[Kevin Wormington]

I’m not using a VIP and I have made some progress by setting a different active 
tag on each node…then upon node failure setting the failed node's tag to active 
on remaining node.  This lets the re-invite pinging work, etc.  It’s almost 
there but the handling of the BYE…they are still sent to the IP of the failed 
node even after re-invite pings so any in-progress calls from the failed node 
are zombie when they hang up until the re-invite ping times out (30 seconds).   
I found an article about initiating a re-invite on the new node with something 
like "opensips-cli -x mi dlg_send_sequential 
callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” but 
that either seems to terminate the call immediately or say the dialog wasn’t 
found.


Thanks,

Kevin
> On Jan 19, 2021, at 8:46 AM, Andy Dierlam  wrote:
> 
> With dialog writing to db that both servers use.   And same tag on both - 
> modparam("dialog", "dlg_sharing_tag", "vip1=active")
> had this working on opensips 2.4 
> 
> thanks
> Andy
> 
> 
> On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington  
> wrote:
> Hi,
> 
> I've been attempting to get a two node active/active setup to work with the 
> v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a 
> proxy that handles all of the NAT/media so either OpenSIPS instance can 
> communicate directly with the user.  
> 
> What I have working so far:
> 
> Registrations and calls work when sent to either node and if you stop 
> OpenSIPS on a node new calls work fine using the other node.
> 
> What I can’t get to work:
> 
> Calls that are already in progress to switch between nodes when one node 
> fails.
> 
> 
> I have messed around with various sharing tags…no tag, same tag, different 
> tags but haven’t had any luck.   I’m guessing that I’m missing something to 
> trigger the remaining node to send re-invites.  Has anyone attempted this 
> type of setup and have any ideas?
> 
> Thanks,
> 
> Kevin
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[OpenSIPS-Users] v3.1 Active/Active maintain active calls on node failure

[Kevin Wormington]

Hi,

I've been attempting to get a two node active/active setup to work with the 
v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a 
proxy that handles all of the NAT/media so either OpenSIPS instance can 
communicate directly with the user.  

What I have working so far:

Registrations and calls work when sent to either node and if you stop OpenSIPS 
on a node new calls work fine using the other node.

What I can’t get to work:

Calls that are already in progress to switch between nodes when one node fails.


I have messed around with various sharing tags…no tag, same tag, different tags 
but haven’t had any luck.   I’m guessing that I’m missing something to trigger 
the remaining node to send re-invites.  Has anyone attempted this type of setup 
and have any ideas?

Thanks,

Kevin
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