Comments for SYSADMINMAN» sysadminman – Asterisk VPS – Trixbox, Elastix, PIAF, A2Billing http://sysadminman.net/blog UK based Asterisk, Trixbox, FreePBX and A2Billing Servers Tue, 02 Mar 2010 17:02:27 +0000 http://wordpress.org/?v=2.9.2 hourly 1 Comment on Getting started with A2Billing – Part 4 Creating a customer and making a call by John http://sysadminman.net/blog/2009/getting-started-with-a2billing-part-4-creating-a-customer-483/comment-page-1#comment-1540 John Tue, 02 Mar 2010 17:02:27 +0000 http://sysadminman.net/blog/?p=483#comment-1540 I have fixed it, finally and can make calls via calling card gateway. Thanks. I have fixed it, finally and can make calls via calling card gateway. Thanks.

]]> Comment on Getting started with A2Billing – Part 5 Importing a ratecard by John http://sysadminman.net/blog/2009/getting-started-with-a2billing-part-5-importing-a-ratecard-512/comment-page-1#comment-1539 John Tue, 02 Mar 2010 16:58:45 +0000 http://sysadminman.net/blog/?p=512#comment-1539 I am trying to do this and I get message: ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'mysql&gt' at line 1 I tried several combinations and none of them worked. Am I typing something wrong? I am trying to do this and I get message:

ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near ‘mysql&gt’ at line 1

I tried several combinations and none of them worked. Am I typing something wrong?

]]> Comment on Getting started with A2Billing – Part 4 Creating a customer and making a call by matt http://sysadminman.net/blog/2009/getting-started-with-a2billing-part-4-creating-a-customer-483/comment-page-1#comment-1538 matt Tue, 02 Mar 2010 08:57:51 +0000 http://sysadminman.net/blog/?p=483#comment-1538 Does the trunk name you've used in a2billing match the name you've used in freepbx? If so I think the only option would be to switch on the debug in a2billing for the agi-conf that you are using and see what the logs think is going on. Does the trunk name you’ve used in a2billing match the name you’ve used in freepbx?

If so I think the only option would be to switch on the debug in a2billing for the agi-conf that you are using and see what the logs think is going on.

]]> Comment on Getting started with A2Billing – Part 4 Creating a customer and making a call by John http://sysadminman.net/blog/2009/getting-started-with-a2billing-part-4-creating-a-customer-483/comment-page-1#comment-1537 John Tue, 02 Mar 2010 08:05:47 +0000 http://sysadminman.net/blog/?p=483#comment-1537 Actually the CWU trunk seems to be working fine with both international and U.S. numbers. In fact, I am able to dial all numbers when calling via X-Lite with 011 prefix (including U.S. numbers); I dial 011+1+212 for New York, and I dial 011+44+XXX for the UK. So, the trunk is working fine, and CWU is accepting 011 prefix, but for some reason, the call is going through when the number is dialed via calling card gateway. I searched online and discovered that I am not the only one having exactly the same problem, but noone seems to know what to do. I think something is wrong with the configuration of a2billing itself, but don't know what. Actually the CWU trunk seems to be working fine with both international and U.S. numbers. In fact, I am able to dial all numbers when calling via X-Lite with 011 prefix (including U.S. numbers); I dial 011+1+212 for New York, and I dial 011+44+XXX for the UK. So, the trunk is working fine, and CWU is accepting 011 prefix, but for some reason, the call is going through when the number is dialed via calling card gateway. I searched online and discovered that I am not the only one having exactly the same problem, but noone seems to know what to do. I think something is wrong with the configuration of a2billing itself, but don’t know what.

]]> Comment on Getting started with A2Billing – Part 4 Creating a customer and making a call by matt http://sysadminman.net/blog/2009/getting-started-with-a2billing-part-4-creating-a-customer-483/comment-page-1#comment-1536 matt Tue, 02 Mar 2010 07:06:49 +0000 http://sysadminman.net/blog/?p=483#comment-1536 What is the prefix for? Are you dialling a number in the US that starts 1212? If so it looks like your calling card is expecting to have 01 on the front of the number but I'm not sure that callwithus is. You could edit the cwu trunk in a2billing and try removing that 01 prefix. What is the prefix for? Are you dialling a number in the US that starts 1212?

If so it looks like your calling card is expecting to have 01 on the front of the number but I’m not sure that callwithus is. You could edit the cwu trunk in a2billing and try removing that 01 prefix.

]]> Comment on Getting started with FreePBX – Part 2 Setting up an extension by Basit http://sysadminman.net/blog/2009/getting-started-with-freepbx-part-2-setting-up-an-extension-353/comment-page-1#comment-1535 Basit Tue, 02 Mar 2010 06:13:22 +0000 http://sysadminman.net/blog/?p=353#comment-1535 Hi, Hope you will be fine. I am running asterisk 1.4.18.1. when i connect to asterisk using Xlite and then try to listen the recorded message it says [Mar 2 16:57:01] NOTICE[5198]: res_odbc.c:530 odbc_obj_connect: Connecting voipodbc [Mar 2 16:57:01] NOTICE[5251]: app_voicemail.c:4838 open_mailbox: Resequencing Mailbox: /var/spool/asterisk/voicemail/brights/220/Old [Mar 2 16:57:17] NOTICE[5198]: res_odbc.c:544 odbc_obj_connect: res_odbc: Connected to voipodbc [Mysql-asterisk] [Mar 2 16:57:49] WARNING[5251]: file.c:644 ast_readaudio_callback: Failed to write frame == Spawn extension (office, *, 1) exited non-zero on 'SIP/203-09da2080' and the message did not play but it says you have message from 203 and then take a long pause and then says press this for new message, It is not playing the message. Any idea. Is this firewall issue? Thanks Hi,
Hope you will be fine. I am running asterisk 1.4.18.1. when i connect to asterisk using Xlite and then try to listen the recorded message it says

[Mar 2 16:57:01] NOTICE[5198]: res_odbc.c:530 odbc_obj_connect: Connecting voipodbc
[Mar 2 16:57:01] NOTICE[5251]: app_voicemail.c:4838 open_mailbox: Resequencing Mailbox: /var/spool/asterisk/voicemail/brights/220/Old
[Mar 2 16:57:17] NOTICE[5198]: res_odbc.c:544 odbc_obj_connect: res_odbc: Connected to voipodbc [Mysql-asterisk]
[Mar 2 16:57:49] WARNING[5251]: file.c:644 ast_readaudio_callback: Failed to write frame
== Spawn extension (office, *, 1) exited non-zero on ‘SIP/203-09da2080′

and the message did not play but it says you have message from 203 and then take a long pause and then says press this for new message, It is not playing the message. Any idea. Is this firewall issue?
Thanks

]]> Comment on Getting started with A2Billing – Part 4 Creating a customer and making a call by John http://sysadminman.net/blog/2009/getting-started-with-a2billing-part-4-creating-a-customer-483/comment-page-1#comment-1533 John Mon, 01 Mar 2010 21:21:55 +0000 http://sysadminman.net/blog/?p=483#comment-1533 No, when I dial the number with the dial prefix 1212 I near the message “The number you trying to reach is currently unavailable”. When I dial the same number with 0111212 I hear the number of minutes left for the call (system calculates the time correctly based on the rate I set up but right after that the message about the number being unavailable follows). Trunk configuration: username=XXXXXXXXX type=friend secret=XXXXXXXXXXXX insecure=very host=sip.callwithus.com dtmfmode=rfc2833 context=from-trunk Asterisk SIP info under Tools shows that Asterisk registered with CWU. Not really sure what's wrong. No, when I dial the number with the dial prefix 1212 I near the message “The number you trying to reach is currently unavailable”. When I dial the same number with 0111212 I hear the number of minutes left for the call (system calculates the time correctly based on the rate I set up but right after that the message about the number being unavailable follows). Trunk configuration:

username=XXXXXXXXX
type=friend
secret=XXXXXXXXXXXX
insecure=very
host=sip.callwithus.com
dtmfmode=rfc2833
context=from-trunk

Asterisk SIP info under Tools shows that Asterisk registered with CWU. Not really sure what’s wrong.

]]> Comment on Getting started with A2Billing – Part 4 Creating a customer and making a call by matt http://sysadminman.net/blog/2009/getting-started-with-a2billing-part-4-creating-a-customer-483/comment-page-1#comment-1531 matt Mon, 01 Mar 2010 20:22:32 +0000 http://sysadminman.net/blog/?p=483#comment-1531 Sounds like maybe your trunk is not working correctly. Does the call appear in the call reports? Sounds like maybe your trunk is not working correctly. Does the call appear in the call reports?

]]> Comment on Getting started with A2Billing – Part 4 Creating a customer and making a call by John http://sysadminman.net/blog/2009/getting-started-with-a2billing-part-4-creating-a-customer-483/comment-page-1#comment-1530 John Mon, 01 Mar 2010 18:41:53 +0000 http://sysadminman.net/blog/?p=483#comment-1530 I have set up everything and it's working except one thing - when I dial the pin number and then phone number, I hear message "The number you trying to reach is currently unavailable". How do I fix this? I have set up everything and it’s working except one thing – when I dial the pin number and then phone number, I hear message “The number you trying to reach is currently unavailable”. How do I fix this?

]]> Comment on Getting started with FreePBX – Part 3 Making external calls by matt http://sysadminman.net/blog/2009/getting-started-with-freepbx-part-3-making-external-calls-372/comment-page-1#comment-1520 matt Mon, 22 Feb 2010 19:28:43 +0000 http://sysadminman.net/blog/?p=372#comment-1520 Hi Jordan, You can easily do this on your Outbound Routes in FreePBX. Check out the help for 'Dial Patterns' as these are what decides which calls are completed using that route. Hi Jordan,

You can easily do this on your Outbound Routes in FreePBX. Check out the help for ‘Dial Patterns’ as these are what decides which calls are completed using that route.

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