Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-15 Thread Nabeel
Here is a list of changes I found: 1) Must build asterisk with ODBC storage enabled for voicemail because using file storage will not store messages in the database. 2) Uncomment the lines *'odbcstorage=asterisk'* and *'odbctable=voicemessages'* in voicemail.conf to enable database storage for m

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-14 Thread Bogdan-Andrei Iancu
Nice ! As a way of helping us (project) back, could synthesize a list with things that did changed since the tutorial was written ? And I will re-generate the tutorial, so other people will benefit from it. Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.ope

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-14 Thread Nabeel
I also found the correct way to deal with the LIMIT problem. Asterisk has a built-in way to deal with this. In file* /etc/asterisk/res_odbc.conf*, the following should be added under [asterisk] : limit => 5 share_connections => no Now everything is working well without problems. Nabeel

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-14 Thread Bogdan-Andrei Iancu
Perfect ! is there any left to be solved, or everything works fine ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 14.07.2016 13:33, Nabeel wrote: Hi Bogdan, I have been able to solve that problem. The issue was that I had asterisk compiled

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-14 Thread Nabeel
Hi Bogdan, I have been able to solve that problem. The issue was that I had asterisk compiled with file storage enabled instead of ODBC storage. I recompiled asterisk with ODBC storage enabled and now database storage is working. Thanks. Nabeel On 14 Jul 2016 11:15 a.m., "Bogdan-Andrei Iancu" w

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-14 Thread Bogdan-Andrei Iancu
Hi Nabeel, 1) that limit should not be necessary, as you should have in DB a single record for each subscriber. If multiple records are returned, it means your data is not correct. 2) in those lines, the "asterisk" and "asteriskcfg" are the names of the odbc connection - I pasted an example

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-14 Thread Bogdan-Andrei Iancu
Hi Nabeel, That means the vmusers and vmaliases do work ok, still the VM storage engine does not. Do you have in voicemail.conf the following: odbcstorage=asteriskrt odbctable=voicemessages Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-04 Thread Nabeel
I have been able to solve the issue of loading numbers without using the voicemail.conf file. After adding the line *'voicemail => odbc,asterisk,vmaliases'* to extconfig.cfg, I removed the suffix " |u " from extensions.conf: exten => _VMR_.,n,Voicemail(${EXTEN:4}*|u*) Now all phone numbers in

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-04 Thread Nabeel
0. The line block was in the default OpenSIPS config, but I agree that it is not in the tutorial so should be removed (for voicemail). 1. I think there is a misunderstanding here. 'limit' is not a column; I am referring to the mysql LIMIT clause: https://dev.mysql.com/doc/refman/5.5/en/select.h

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-04 Thread Bogdan-Andrei Iancu
Thank you Nabeel, The number you added in voicemail file - does it exist in the sipuser/subscriber table ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04.07.2016 11:38, Nabeel wrote: Hi Bogdan, I just added the column to the view by ad

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-04 Thread Bogdan-Andrei Iancu
Hi, 0. The cfg block you mentioned as removed does not exists in the cfg as per tutorial. 1. the "limit" column does not exist in the sipusers as per tutorial, so it might have been added in newer asterisk versions; not sure what is its meaning, but if setting it to 1 makes asterisk happy, i

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-04 Thread Nabeel
Hi Bogdan, I just added the column to the view by adding "NULL AS `callbackextension`" to the SQL view definition. I haven't linked the column to the subscriber column, so this may not be the correct definition. However, it got rid of the error. About the voicemail.conf file, when I attempted to

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-04 Thread Bogdan-Andrei Iancu
Hi, the voicemail.conf file exists in almost all asterisk versions. But if you use the odbc storage for voicemail, you do not need this file at all. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 02.07.2016 15:41, Nabeel wrote: In the lates

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-04 Thread Bogdan-Andrei Iancu
Hi, What is the definition you used for this new column ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 02.07.2016 05:29, Nabeel wrote: In the last error message,/'//callbackextension = ?' /suggested that this column is missing from the

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-04 Thread Bogdan-Andrei Iancu
Hi, This kind of ordering is valid in older versions of Asterisk. Maybe not anymore in the newer versions. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 02.07.2016 04:23, Nabeel wrote: The tutorial contains a mistake where the priority order

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-03 Thread Nabeel
Hi Samy, Point 1 I cant imagine how those lines possibly relate to no media error in > asterisk, I guess it depends on your config setup. In point 1 I was referring to this error: WARNING[17112] res_odbc.c: SetConnectAttr (Txn isolation) returned an > error: HY000: [MySQL][ODBC 5.2(w) Driver]Yo

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-03 Thread SamyGo
Hi Nabeel, Point 1 I cant imagine how those lines possibly relate to no media error in asterisk, I guess it depends on your config setup. The logical answer to your point 2 would be Asterisk realtime. However this is not going to be as staraight forward as making asterisk use subscriber table for

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-03 Thread Nabeel
The last error message has been solved by removing the following lines from opensips.cfg: if (!db_does_uri_exist()) { >send_reply("420","Bad Extension"); >exit; >} > >t_newtran(); >t_reply("480",

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-02 Thread Nabeel
In the latest version of Asterisk, there is a new file voicemail.conf which must be configured correctly for voicemail, but the tutorial does not mention this file at all. Please let me know how to configure this file for integration with OpenSIPS. Nabeel __

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-01 Thread Nabeel
In the last error message,* '**callbackextension = ?' *suggested that this column is missing from the sipusers mysql view. So I added this column to the view and now that error has been resolved. Only the following error remains now: [Jul 2 03:25:48] WARNING[19330][C-0005]: app.c:1633 > __ast

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-01 Thread Nabeel
The issue in my last Email has solved the error about missing extension. Now the following errors remain: [Jul 2 02:29:18] WARNING[18226][C-0001]: res_config_odbc.c:117 > custom_prepare: SQL Prepare failed![SELECT * FROM sipusers WHERE host = ? > AND callbackextension = ? AND port = ?] > [Jul

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-01 Thread Nabeel
The tutorial contains a mistake where the priority ordering in extensions.conf should start with 1, not n: ; Voicemail > exten => _VMR_.,1,Ringing > exten => _VMR_.,n,Wait(1) > exten => _VMR_.,n,Answer > exten => _VMR_.,n,Wait(1) > exten => _VMR_.,n,Voicemail(${EXTEN:4}|u) > exten => _VMR_.,n,Hang

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-07-01 Thread Nabeel
Hi, Adding 'limit 1' or 'limit 5' to the supusers mysql view resolves part of the error, but I don't understand why that is and whether this is correct for the setup. Maybe something to do with connection pooling? Now the following errors remain: [Jun 30 01:07:53] NOTICE[17067][C-] chan_

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-06-30 Thread Bogdan-Andrei Iancu
Hi Nabeel, The "sipusers" mysql view (as per http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8#toc7 ) has both the name and host fields - not sure why that query may fail. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-soluti

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-06-29 Thread Nabeel
Hi Bogdan, I was able to install the latest versions of Asterisk (13.1) and Opensips (2.3) according to the tutorial, but when attempting to leave a voicemail I get the following errors: > [Jun 30 01:07:53] NOTICE[17067][C-] chan_sip.c: Call from > '+447867958678' (162.249.6.206:12221) t

Re: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-06-14 Thread Bogdan-Andrei Iancu
Hi Nabeel, We will update the tutorial for 2.2, but it should still match. Give it a try and if you hit issues, just let me know. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 12.06.2016 10:18, Nabeel wrote: Hi, I will be following this t

[OpenSIPS-Users] OpenSIPS and Asterisk integration versions

2016-06-12 Thread Nabeel
Hi, I will be following this tutorial to integrate OpenSIPS and Asterisk: http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8 The tutorial mentions the use of OpenSIPS version 1.8 and Asterisk version 1.8. I would like to know if I can use the latest versions of OpenS