Re: [asterisk-users] Asterisk - Trixbox
They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
I disabled that last number's registration and moved to a new number (to test each number individually without the sip debugging from the others). I waited maybe 5 minutes and I restarted Asterisk to ensure the other side was done with whatever it was doing. I called the second number (8152641125) and the first number (8159911010) shows up as the peer. Not only that, but with this number, there's no compatible codecs. I ensured that both entries in sip.conf were the same other than things that needed to be different such as username. I even had that entry have allow=all. I still get the codec error. http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is the peer issue whereas 42 is the codec issue. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Do you know enough about Trixbox to tell me where they need to fix their misconfiguration, or is it a Trixbox design flaw? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Yes, they should fix this on their side, otherwise DID routing will not work. If you don't need it, you just need to create a DID entry for any/all or any/any, I cannot remember which it is right now, but it should be apparent when you look at it. The s extension is only used when no DID or extension is received. Thanks, Steve Totaro On Tue, Feb 10, 2009 at 9:49 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
It's a local CLEC, Essex Telcom. The burden does lie with them, but I doubt they'll fix it since if you provision a grandstream, it works just fine. I have a total of 5 numbers with them. Two are on the server that is experiencing issues. Another is on a different server with no issues. The remaining two aren't provisioned anywhere. I'm going to be adding another number shortly. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@totarotechnologies.com Sent: Tuesday, February 10, 2009 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox How many accounts do you have? If just one, then a single peer should be fine but they should be sending the destination exten as a DID, obviously they are not. I think the burden of fixing it lies with them? What carrier is this? On Tue, Feb 10, 2009 at 10:09 AM, Mike Hammett asterisk-us...@ics-il.net wrote: I disabled that last number's registration and moved to a new number (to test each number individually without the sip debugging from the others). I waited maybe 5 minutes and I restarted Asterisk to ensure the other side was done with whatever it was doing. I called the second number (8152641125) and the first number (8159911010) shows up as the peer. Not only that, but with this number, there's no compatible codecs. I ensured that both entries in sip.conf were the same other than things that needed to be different such as username. I even had that entry have allow=all. I still get the codec error. http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is the peer issue whereas 42 is the codec issue. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Yeah. They were running a Clarent switch and that's the one that came down. They also had\have a Coppercom switch. The Clarent was old, though I really didn't have any problems with it. I could never get the Coppercom to work with Asterisk (though I'm an expert at neither) and their tech support told my carrier to fly a kite when we were having T38 issues. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@totarotechnologies.com Sent: Monday, February 02, 2009 9:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Your carrier is running Trixbox? That is scary. Anyways, they are obviously routing calls to the wrong machine. If your side worked properly before and now does not, then they have to explain why. That would be my stance anyways. Thanks, Steve On Mon, Feb 2, 2009 at 10:18 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Mike Hammett asterisk-us...@ics-il.net Sent: Thursday, January 29, 2009 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Mike Hammett asterisk-us...@ics-il.net Sent: Thursday, January 29, 2009 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Trixbox
My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attrafax
Has anyone had any luck with Attrafax? I'm looking to use it as the T.38 gateway (PRI in, T.38 out). -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors
Thanks for the help. I still had a misconfiguration in my res_odbc.conf, but I figured it out and it appears my voicemail storage is working. I haven't had a chance to get to the phone on the extension I'm using for it. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, March 11, 2008 10:10 PM Subject: Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors [EMAIL PROTECTED] asterisk]# cat /etc/odbc.ini [astrealtime] Description = MySQL ODBC For Asterisk Trace = Yes TraceFile = /tmp/odbc.log Driver = MySQL Server = localhost User= astrealtime Password= Database= asterisk Socket = /var/lib/mysql/mysql.sock [EMAIL PROTECTED] asterisk]# cat /etc/odbcinst.ini [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc.so Setup = /usr/lib/libodbcmyS.so UsageCount = 4 [MySQL ODBC 3.51 Driver] Description = ODBC 3.51 for MySQL DRIVER = /usr/lib/libmyodbc3.so SETUP = /usr/lib/libmyodbc3S.so UsageCount = 4 [EMAIL PROTECTED] asterisk]# isql astrealtime astrealtime [ISQL]ERROR: Could not SQLConnect [EMAIL PROTECTED] asterisk]# locate libmyodbc /usr/lib64/libmyodbc3-3.51.12.so /usr/lib64/libmyodbc3.so /usr/lib64/libmyodbc3_r-3.51.12.so /usr/lib64/libmyodbc3_r.so [EMAIL PROTECTED] asterisk]# locate libodbcmyS /usr/lib/libodbcmyS.a /usr/lib/libodbcmyS.so /usr/lib/libodbcmyS.so.1 /usr/lib/libodbcmyS.so.1.0.0 /usr/lib64/libodbcmyS.a /usr/lib64/libodbcmyS.so /usr/lib64/libodbcmyS.so.1 /usr/lib64/libodbcmyS.so.1.0.0 [EMAIL PROTECTED] asterisk]# rpm -qa | grep odbc mysql-connector-odbc-3.51.12-2.2 [EMAIL PROTECTED] asterisk]# rpm -qa | grep ODBC unixODBC-2.2.11-7.1 unixODBC-2.2.11-7.1 unixODBC-devel-2.2.11-7.1 unixODBC-devel-2.2.11-7.1 [EMAIL PROTECTED] asterisk]# mysql -u astrealtime -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 484 to server version: 5.0.22 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use asterisk; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql show tables; ++ | Tables_in_asterisk | ++ | cdr| | extensions_table | | iax| | queue_members | | queues | | sip| | voicemail_messages | | voicemail_users| ++ 8 rows in set (0.00 sec) -- Executing [EMAIL PROTECTED]:1] VoiceMail(SIP/9826-ac087500, [EMAIL PROTECTED]|u) in new stack [Mar 11 21:29:36] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:29:36] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:29:36] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:29:36] WARNING[26144]: app_voicemail.c:1122 retrieve_file: Failed to obtain database object for 'mysql'! [Mar 11 21:29:36] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:29:36] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:29:36] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:29:36] WARNING[26144]: app_voicemail.c:1122 retrieve_file: Failed to obtain database object for 'mysql'! [Mar 11 21:29:36] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:29:36] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:29:36] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:29:36] WARNING[26144]: app_voicemail.c:1122 retrieve_file: Failed to obtain database object for 'mysql'! -- SIP/9826-ac087500 Playing 'vm-theperson' (language 'en') -- SIP/9826-ac087500 Playing 'digits/2' (language 'en') -- SIP/9826-ac087500 Playing 'digits/0' (language 'en') -- SIP/9826-ac087500 Playing 'digits/0' (language 'en') -- SIP/9826-ac087500 Playing 'vm-isunavail' (language 'en') -- SIP/9826-ac087500 Playing 'vm-intro' (language 'en') [Mar 11 21:29:46] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21
[asterisk-users] sip.conf help, inbound calls fall to last specified context
First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to remove any source of typo. [EMAIL PROTECTED] asterisk]# cat sip.conf [general] context=default port=5060 canreinvite=no ;register = 8157582715::[EMAIL PROTECTED] ; ottos 815-758-2715 register = 8157879826::[EMAIL PROTECTED] ; ottos 815-787-9826 ;register = 8159092441::[EMAIL PROTECTED] ; RWest 815-909-2441 ;register = 8159092443::[EMAIL PROTECTED] ; RWest 815-909-2443 ;- REALTIME SUPPORT ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read realtime.txt and extconfig.txt in the /doc directory of the ; source code. ; rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) ;rtsavesysname=yes ; Save systemname in realtime database at registration ; Default= no ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. ; If not present, defaults to 'yes'. ;rtautoclear=yes; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|seconds) ; If set to yes, when the registration expires, the friend will ; vanish from the configuration until requested again. If set ; to an integer, friends expire within this number of seconds ; instead of the registration interval. ;ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage [8157582715] type=friend accountcode=2 context=ottos secret= username=2715 fromuser=8157582715 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com canreinvite=no disallow=all allow=ulaw allow=alaw ;qualify=yes [8159092441] type=friend accountcode=12 context=rwest secret= username=2441 fromuser=8159092441 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com canreinvite=no disallow=all allow=ulaw allow=alaw ;qualify=yes [8159092443] type=friend accountcode=12 context=rwest secret= username=2441 fromuser=8159092443 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com canreinvite=no disallow=all allow=ulaw allow=alaw ;qualify=yes [8157879826] type=friend ;accountcode=2 context=ics secret= username=9826 fromuser=8157879826 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com ;canreinvite=no ;disallow=all ;allow=ulaw -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf help, inbound calls fall to last specified context
Updated with a smaller sip.conf that also doesn't work right. [EMAIL PROTECTED] asterisk]# cat sip.conf [general] port=5060 canreinvite=no rtcachefriends=yes disallow=all allow=ulaw allow=alaw register = 8157879826::[EMAIL PROTECTED] ; ottos 815-787-9826 register = 8159092443::[EMAIL PROTECTED] ; RWest 815-909-2443 [8157879826] type=friend accountcode=2 context=ics secret= username=9826 fromuser=8157589826 insecure=very host=voip.essex1.com fromdomain=voip.essex1.com [8159092443] type=friend accountcode=12 context=rwest secret= username=2441 fromuser=8159092443 insecure=very host=63.175.151.3 ;voip.essex1.com fromdomain=63.175.151.3 ;voip.essex1.com -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Thursday, March 13, 2008 9:13 AM Subject: [asterisk-users] sip.conf help,inbound calls fall to last specified context First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to remove any source of typo. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mail Server
I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail Server
Through help from people on the lists and then further investigation based on those results, here is what I did. 1) I set the office to a statically assigned IP instead of from the pool. 2) I made an A entry on one of my domains aiur.ics-il.net (where aiur is the machine name). 3) I added aiur.ics-il.net directly after 127.0.0.1 in the /etc/hosts file (copied below). 4) I set the from email address (serveremail) in /etc/asterisk/voicemail.conf to something at the domain I created ([EMAIL PROTECTED]). 5) Presto! [EMAIL PROTECTED] ~]# cat /etc/hosts # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1 aiur.ics-il.net Aiurlocalhost.localdomain localhost ::1 localhost6.localdomain6 localhost6 -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, March 13, 2008 4:04 PM Subject: [asterisk-users] Mail Server I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail Server
I am the ISP. ;-) I'll have to look into that smarthost deal as there is no reverse DNS at this time (my upstream's server times out). -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Erik Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 13, 2008 4:25 PM Subject: Re: [asterisk-users] Mail Server On Thu, Mar 13, 2008 at 4:04 PM, Mike Hammett [EMAIL PROTECTED] wrote: I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT. Does your ISP provide an SMTP server you can use? If so, it's usually easiest to set that up as a smarthost and tell sendmail to send through that server. If this isn't an option, you need to make sure that your asterisk server has a valid publicly-available DNS record (and reverse DNS). That's most likely the reason the remote server is rejecting these emails. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple clients registering on same definition in Realtime
I was going to setup my extension on my employee's phone so he could answer calls as well as myself. I noticed that once he registered, I could no longer receive calls on my own phone. Is this a limitation of Realtime or something else in Asterisk? I've had multiple devices register to the same definition somewhere else before in Asterisk. If I can't do it that way, I'm thinking of having his phone register as some other, new extension (in addition to himself) and just have calls to my extension ring that new extension as well. I'd also have that new extension's voicemail point to my box, therefore he can check my voicemails as well. He has a Cisco 7960 and currently all voicemails (whether you enter sales, support, etc.) all dump into my box. Perhaps I'll divide that up as well to take advantage of the 6 lines on his phone. Comments or alternative suggestions? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors
: /var/spool/asterisk/voicemail/ics/200/tmp/oDVLVZ format: wav, 0x122f3740 -- User hung up [Mar 11 21:30:25] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:30:25] WARNING[26144]: app_voicemail.c:1233 message_exists: Failed to obtain database object for 'mysql'! [Mar 11 21:30:25] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:30:25] WARNING[26144]: app_voicemail.c:1263 delete_file: Failed to obtain database object for 'mysql'! [Mar 11 21:30:25] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:30:25] WARNING[26144]: app_voicemail.c:1400 store_file: Failed to obtain database object for 'mysql'! [Mar 11 21:30:25] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:30:25] WARNING[26144]: app_voicemail.c:1122 retrieve_file: Failed to obtain database object for 'mysql'! [Mar 11 21:30:25] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:30:25] WARNING[26144]: app_voicemail.c:2277 messagecount: Failed to obtain database object for 'mysql'! [Mar 11 21:30:25] NOTICE[26144]: res_odbc.c:508 odbc_obj_connect: Connecting mysql [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:519 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Mar 11 21:30:25] WARNING[26144]: res_odbc.c:444 ast_odbc_request_obj: Failed to connect to mysql [Mar 11 21:30:25] WARNING[26144]: app_voicemail.c:2277 messagecount: Failed to obtain database object for 'mysql'! == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/9826-ac087500' in macro 'stdexten' == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/9826-ac087500' -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, February 22, 2008 7:28 PM Subject: [asterisk-users] MySQL Voicemail Storage Questions\Errors I am running CentOS 5 with Asterisk 1.4.14. I am trying to setup storage of voicemail messages into MySQL. It is my understanding that I can only do this via ODBC. I installed per http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured and built Asterisk, using menuconfig to turn on ODBC voicemail storage. Here is the output of some config files: [EMAIL PROTECTED] asterisk]# cat /etc/odbcinst.ini # Example driver definitinions # # # Included in the unixODBC package #[PostgreSQL] #Description= ODBC for PostgreSQL #Driver = /usr/lib/libodbcpsql.so #Setup = /usr/lib/libodbcpsqlS.so #FileUsage = 1 # Driver from the MyODBC package # Setup from the unixODBC package [MySQL] Description = ODBC for MySQL Driver = /usr/lib64/libmyodbc3.so Setup = /usr/lib64/libodbcmyS.so FileUsage = 1 You have new mail in /var/spool/mail/root [EMAIL PROTECTED] asterisk]# cat /etc/odbc.ini [astrealtime] Description = Asterisk realtime FUNC_ODBC access Driver = MySQL Socket = /var/lib/mysql/mysql.sock Server = localhost User= astrealtime Pass= Database= asterisk Option = 3 [EMAIL PROTECTED] asterisk]# cat /etc/asterisk
Re: [asterisk-users] Cisco 7960 SIP Upgrade
As expected, Jim took care of me WRT the Cisco upgrade. It is now far more usable than when it was SCCP... I gave up on trying to get SCCP working in Asterisk after upgrading to 1.4 from 1.0. Due to his generosity, I feel I owe him to recommend his termination\origination services. The one or two times I've had any issue, he has been quick to respond and took care of me. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Sigma Networks [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 04, 2008 12:34 PM Subject: Re: [asterisk-users] Cisco 7960 SIP Upgrade Mike Hammett wrote: I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I know you are a very happy customer of Sigma Networks ( :-) )... I'd be happy to upgrade the phone to 8.3.3SR2 for you. Jim ph: 408-701-9929 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Coppercom and Asterisk
*bump* -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 21, 2008 11:55 AM Subject: Re: [asterisk-users] Coppercom and Asterisk I put that in, but it appears that it is trying to contact the private IP address of their SIP server. I have successfully registered to this server from over the public Internet using an Innomedia ATA. [Feb 21 11:49:18] NOTICE[4608]: chan_sip.c:7364 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER Retransmitting #1 (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Retransmitting #2 (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Retransmitting #3 (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER Retransmitting #4 (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Aiur*CLI -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 21, 2008 1:53 AM Subject: Re: [asterisk-users] Coppercom and Asterisk In the [general] section, put: register = 8159093010:[EMAIL PROTECTED] Then add a SIP peer for the outbound proxy. Something like: [essex1_outbound] fromdomain=proxy.essex1.com host=proxy.essex1.com port=5060 insecure=very username=8159093010 secret=X type=peer qualify=no canreinvite=no dtmfmode=rfc2833 disallow=all allow=ulaw The first one is needed for the registrations, and the second one is needed to answer 407 proxy challenges. Mike Hammett wrote: My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - X No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use outbound Proxy -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation
[asterisk-users] Cisco 7960 SIP Upgrade
I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Coppercom and Asterisk
I was doing it because of the volume on the server. It is very easy to miss a message or 10 or 100 on a list of this traffic. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 04, 2008 10:30 AM Subject: Re: [asterisk-users] Coppercom and Asterisk On Tuesday 04 March 2008 09:45:38 Mike Hammett wrote: *bump* If people don't know, they don't know. There is no need to repost your query 10 days later. Not that many more people have signed up, and those who have signed up are unlikely to be able to answer your question. The only thing that this does is serve to annoy the rest of the people on the list. Please do not do it again. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP Upgrade
That I am. I'll contact you off list. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Sigma Networks [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 04, 2008 12:34 PM Subject: Re: [asterisk-users] Cisco 7960 SIP Upgrade Mike Hammett wrote: I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I know you are a very happy customer of Sigma Networks ( :-) )... I'd be happy to upgrade the phone to 8.3.3SR2 for you. Jim ph: 408-701-9929 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Coppercom and Asterisk
register = [EMAIL PROTECTED]:X:[EMAIL PROTECTED] [8159093010] fromdomain=proxy.essex1.com host=proxy.essex1.com port=5060 insecure=very username=8159093010 secret=X type=peer qualify=no canreinvite=no dtmfmode=rfc2833 disallow=all allow=ulaw outboundproxy=proxy.essex1.com [Feb 28 07:44:52] NOTICE[9409]: chan_sip.c:7364 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]@proxy.essex1.com' timed out, trying again (Attempt #1) REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 63.164.210.14:5060: REGISTER sip:proxy.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK24661abe;rport From: sip:[EMAIL PROTECTED];tag=as16c1714c To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Aiur*CLI --- SIP read from 63.164.210.14:5060 --- SIP/2.0 423 Interval Too Brief To: sip:[EMAIL PROTECTED];tag=ddcdjfgdeigdhifj-bibgaceacb From: sip:[EMAIL PROTECTED];tag=as16c1714c Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK24661abe Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Expires: 120 Min-Expires: 900 Content-Length: 0 - --- (9 headers 0 lines) --- -- Got SIP response 423 Interval Too Brief back from 63.164.210.14 Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER [Feb 28 07:45:12] NOTICE[9409]: chan_sip.c:7364 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]@proxy.essex1.com' timed out, trying again (Attempt #2) REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 63.164.210.14:5060: REGISTER sip:proxy.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK7db9ed82;rport From: sip:[EMAIL PROTECTED];tag=as4a12e1ea To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Aiur*CLI --- SIP read from 63.164.210.14:5060 --- SIP/2.0 423 Interval Too Brief To: sip:[EMAIL PROTECTED];tag=ejhgidfdeiidhifj-bacgaceacb From: sip:[EMAIL PROTECTED];tag=as4a12e1ea Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK7db9ed82 Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER Expires: 120 Min-Expires: 900 Content-Length: 0 - --- (9 headers 0 lines) --- -- Got SIP response 423 Interval Too Brief back from 63.164.210.14 Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Wednesday, February 20, 2008 4:52 PM Subject: [asterisk-users] Coppercom and Asterisk My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - X No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use outbound Proxy -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Coppercom and Asterisk
It is, however, heavily trafficked and easy for someone to miss an email. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Benny Amorsen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 25, 2008 3:44 AM Subject: Re: [asterisk-users] Coppercom and Asterisk Mike Hammett [EMAIL PROTECTED] writes: *bump* This is not some silly forum. *plonk* /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Coppercom and Asterisk
I thought it was odd, but I've had other devices work properly with that information. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Chris Bagnall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, February 24, 2008 8:38 PM Subject: Re: [asterisk-users] Coppercom and Asterisk Proxy - sip.essex1.com (10.1.3.2) Isn't it a bit unusual for their proxy to be given to you as an RFC1918 address? Unless you're on their LAN of course... Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Coppercom and Asterisk
*bump* -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Wednesday, February 20, 2008 4:52 PM Subject: [asterisk-users] Coppercom and Asterisk My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - X No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use outbound Proxy -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL Voicemail Storage Questions\Errors
|| | mailboxcontext | varchar(80) | YES | | NULL|| | recording | longblob| YES | | NULL|| ++-+--+-+-++ 11 rows in set (0.00 sec) mysql exit; Bye Here is an example of the errors I'm getting (ignore No route to destination, those phones just aren't on): -- Executing Goto(SIP/2441-ac047f90, rwest|815XXX|1) -- Goto (rwest,815XXX,1) -- Executing NoOp(SIP/2441-ac047f90, CallerID is WIRELESS CALLER XXX) -- Executing Dial(SIP/2441-ac047f90, SIP/rwest200SIP/rwest201SIP/rwest202SIP/rwest203|15) [Feb 22 18:14:42] WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database object for 'asterisk'! -- Called rwest200 [Feb 22 18:14:42] WARNING[21149]: app_dial.c: dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Feb 22 18:14:42] WARNING[21149]: app_dial.c: dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Feb 22 18:14:42] WARNING[21149]: app_dial.c: dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Feb 22 18:14:42] NOTICE[21068]: chan_sip.c:12414 handle_response_peerpoke: Peer 'rwest200' is now Reachable. (37ms / 2000ms) -- SIP/rwest200-19612180 is ringing [Feb 22 18:14:53] WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database object for 'asterisk'! -- Nobody picked up in 15000 ms [Feb 22 18:14:57] NOTICE[21149]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/rwest200-19612180' not posted -- Executing BackGround(SIP/2441-ac047f90, /var/lib/asterisk/sounds/rwestgreeting) -- SIP/2441-ac047f90 Playing '/var/lib/asterisk/sounds/rwestgreeting' (language 'en') [Feb 22 18:15:04] WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database object for 'asterisk'! == CDR updated on SIP/2441-ac047f90 -- Executing Voicemail(SIP/2441-ac047f90, [EMAIL PROTECTED]|u) [Feb 22 18:15:09] WARNING[21149]: app_voicemail.c:1122 retrieve_file: Failed to obtain database object for 'asterisk'! [Feb 22 18:15:09] WARNING[21149]: app_voicemail.c:1122 retrieve_file: Failed to obtain database object for 'asterisk'! [Feb 22 18:15:09] WARNING[21149]: app_voicemail.c:1122 retrieve_file: Failed to obtain database object for 'asterisk'! -- SIP/2441-ac047f90 Playing 'vm-theperson' (language 'en') -- SIP/2441-ac047f90 Playing 'digits/2' (language 'en') -- SIP/2441-ac047f90 Playing 'digits/0' (language 'en') -- SIP/2441-ac047f90 Playing 'digits/0' (language 'en') -- SIP/2441-ac047f90 Playing 'vm-isunavail' (language 'en') -- SIP/2441-ac047f90 Playing 'vm-intro' (language 'en') [Feb 22 18:15:15] WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database object for 'asterisk'! [Feb 22 18:15:20] WARNING[21149]: app_voicemail.c:1187 last_message_index: Failed to obtain database object for 'asterisk'! -- SIP/2441-ac047f90 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/rwest/200/tmp/me15Bl format: wav, 0x19652c30 [Feb 22 18:15:26] WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database object for 'asterisk'! -- User hung up [Feb 22 18:15:26] WARNING[21149]: app_voicemail.c:1233 message_exists: Failed to obtain database object for 'asterisk'! [Feb 22 18:15:26] WARNING[21149]: app_voicemail.c:1263 delete_file: Failed to obtain database object for 'asterisk'! [Feb 22 18:15:26] WARNING[21149]: app_voicemail.c:1400 store_file: Failed to obtain database object for 'asterisk'! [Feb 22 18:15:26] WARNING[21149]: app_voicemail.c:1122 retrieve_file: Failed to obtain database object for 'asterisk'! [Feb 22 18:15:26] WARNING[21149]: app_voicemail.c:2277 messagecount: Failed to obtain database object for 'asterisk'! [Feb 22 18:15:26] WARNING[21149]: app_voicemail.c:2277 messagecount: Failed to obtain database object for 'asterisk'! == Spawn extension (rwest, 300, 1) exited non-zero on 'SIP/2441-ac047f90' -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors
It was my understanding that voicemail.conf referenced MySQL and not asterisk. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 22, 2008 6:56 PM Subject: Re: [asterisk-users] MySQL Voicemail Storage Questions\Errors On Friday 22 February 2008 18:28:56 Mike Hammett wrote: --snip-- [asterisk] enabled = no dsn = asterisk ;username = myuser ;password = mypass pre-connect = yes --snip-- WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database object for 'asterisk'! What does enabled mean to you? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Coppercom and Asterisk
I put that in, but it appears that it is trying to contact the private IP address of their SIP server. I have successfully registered to this server from over the public Internet using an Innomedia ATA. [Feb 21 11:49:18] NOTICE[4608]: chan_sip.c:7364 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER Retransmitting #1 (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Retransmitting #2 (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Retransmitting #3 (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER Retransmitting #4 (no NAT) to 10.1.3.2:5060: REGISTER sip:sip.essex1.com SIP/2.0 Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport From: sip:[EMAIL PROTECTED];tag=as58a684a6 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Aiur*CLI -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 21, 2008 1:53 AM Subject: Re: [asterisk-users] Coppercom and Asterisk In the [general] section, put: register = 8159093010:[EMAIL PROTECTED] Then add a SIP peer for the outbound proxy. Something like: [essex1_outbound] fromdomain=proxy.essex1.com host=proxy.essex1.com port=5060 insecure=very username=8159093010 secret=X type=peer qualify=no canreinvite=no dtmfmode=rfc2833 disallow=all allow=ulaw The first one is needed for the registrations, and the second one is needed to answer 407 proxy challenges. Mike Hammett wrote: My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - X No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use outbound Proxy -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Coppercom and Asterisk
My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - X No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use outbound Proxy -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Call Recording
Are there any tricks to getting combine_wave to make? [EMAIL PROTECTED] combine_wave-0.3]# ls -al total 84 drwxr-xr-x 2 root root 4096 Jan 15 10:54 . drwxr-x--- 6 root root 4096 Jan 15 10:54 .. -rw-r--r-- 1 root root 351 Oct 6 2005 CHANGES -rw-r--r-- 1 root root 1123 Oct 6 2005 combine_wave-0.3.lsm -rw-r--r-- 1 root root 23280 Oct 6 2005 combine_wave.c -rw-r--r-- 1 root root 449 Oct 6 2005 combine_wave.h -rw-r--r-- 1 root root 1048 Oct 6 2005 combine_wave.man -rw-r--r-- 1 root root 17976 Oct 6 2005 LICENSE -rw-r--r-- 1 root root 459 Oct 6 2005 Makefile -rw-r--r-- 1 root root 341 Oct 6 2005 README -rw-r--r-- 1 root root 762 Oct 6 2005 wave_header.h [EMAIL PROTECTED] combine_wave-0.3]# nano README [EMAIL PROTECTED] combine_wave-0.3]# make gcc -O2 -Wall -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -c combine_wave.c combine_wave.c: In function ârunning_infoâ: combine_wave.c:22: error: missing terminating character combine_wave.c:24: error: âbâ undeclared (first use in this function) combine_wave.c:24: error: (Each undeclared identifier is reported only once combine_wave.c:24: error: for each function it appears in.) combine_wave.c:24: error: expected â)â before âtogglesâ combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: missing terminating character combine_wave.c:36: error: expected â;â before â}â token combine_wave.c: In function âusageâ: combine_wave.c:42: error: missing terminating character combine_wave.c:44: error: âcombine_waveâ undeclared (first use in this function) combine_wave.c:44: error: âaâ undeclared (first use in this function) combine_wave.c:44: error: âdâ undeclared (first use in this function) combine_wave.c:44: error: expected â]â before âmilliâ combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: expected â)â before ânâ combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: missing terminating character combine_wave.c:62: error: expected â;â before â}â token combine_wave.c: In function âstrsaveâ: combine_wave.c:71: warning: implicit declaration of function âstrlenâ combine_wave.c:71: warning: incompatible implicit declaration of built-in function âstrlenâ combine_wave.c:73: warning: implicit declaration of function âstrcpyâ combine_wave.c:73: warning: incompatible implicit declaration of built-in function âstrcpyâ combine_wave.c: In function âmainâ: combine_wave.c:604: warning: incompatible implicit declaration of built-in function âstrcpyâ combine_wave.c:991: warning: implicit declaration of function âmemcpyâ combine_wave.c:991: warning: incompatible implicit declaration of built-in function âmemcpyâ make: *** [combine_wave.o] Error 1 - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Steve Johnson [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 14, 2008 10:51 AM Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording You might take a few ideas from this combine.sh script which works for me. It uses the combine_wave program from http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame program to convert to mp3. It converts the entire directory /var/spool/asterisk/monitor/*-in.wav files to mp3 where the mp3 file doesn't already exist. S. File: combine.sh --- #!/bin/sh cd /var/spool/asterisk/monitor for f in *-in.wav do in=$f out=`echo $f | sed -e 's/-in.wav/-out.wav/'` tmpwav=`echo $f | sed -e 's/-in.wav/-both.wav/'` mp3=`echo $f | sed -e 's/-in.wav/.mp3/'` if [ -e $mp3 ] then continue fi # combine the two tracks into one stereo file /usr/local/bin/combine_wave -l $in -r $out -o $tmpwav 2/dev/null /usr/bin/lame --silent -h -b 96 $tmpwav $mp3
Re: [asterisk-users] Asterisk 1.4 Call Recording
I'm a newb when it comes to patch. I have a combine_wave-0.3.orig and a
combine_wave-0.3 directory. This is what I get:
[EMAIL PROTECTED] ~]# patch combine_wave-0.3.patch
can't find file to patch at input line 4
Perhaps you should have used the -p or --strip option?
The text leading up to this was:
--
|diff -Naur combine_wave-0.3.orig/combine_wave.c
combine_wave-0.3/combine_wave.c
|--- combine_wave-0.3.orig/combine_wave.c 2005-10-06
14:44:10.0 +0200
|+++ combine_wave-0.3/combine_wave.c2007-10-05 21:02:17.0 +0200
--
File to patch:
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 15, 2008 11:19 AM
Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording
Hi Mike,
On Tue, 2008-01-15 at 10:57 -0600, Mike Hammett wrote:
Are there any tricks to getting combine_wave to make?
Patch attached. Builds fine with patch on Fedora 8.
Regards,
Patrick
diff -Naur combine_wave-0.3.orig/combine_wave.c
combine_wave-0.3/combine_wave.c
--- combine_wave-0.3.orig/combine_wave.c 2005-10-06 14:44:10.0
+0200
+++ combine_wave-0.3/combine_wave.c 2007-10-05 21:02:17.0 +0200
@@ -19,8 +19,8 @@
void running_info()
{
-fprintf(stderr,\
-RUNNNING COMMANDS
+fprintf(stderr,
+RUNNNING COMMANDS\n\
b toggles move both channels / move right channel delay mode.\n\
ESC exits.\n\
'z' 'x' 1 sample forward / backward.\n\
@@ -39,8 +39,8 @@
void usage()
{
-fprintf(stderr,\
-Usage:
+fprintf(stderr,
+Usage:\n\
combine_wave [-a] [-d milli seconds delay right channel relative to
left]\n\
[-e samples delay right channel relative to left]\n\
[-k] -l filename_left [-m] -o output_filename -r filename_right [s start
seek offset].\n\
diff -Naur combine_wave-0.3.orig/combine_wave.h
combine_wave-0.3/combine_wave.h
--- combine_wave-0.3.orig/combine_wave.h 2005-10-06 14:44:10.0
+0200
+++ combine_wave-0.3/combine_wave.h 2007-10-05 21:02:52.0 +0200
@@ -5,6 +5,7 @@
#include unistd.h
#include stdio.h
#include stdlib.h
+#include string.h
#include signal.h
#include errno.h
diff -Naur combine_wave-0.3.orig/Makefile combine_wave-0.3/Makefile
--- combine_wave-0.3.orig/Makefile 2005-10-06 14:44:10.0 +0200
+++ combine_wave-0.3/Makefile 2007-10-05 21:00:43.0 +0200
@@ -6,13 +6,13 @@
CFLAGS = -O2 -Wall -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
.c.o:
- gcc $(CFLAGS) -c $
+ $(CC) $(CFLAGS) -c $
OBJECT =\
combine_wave.o
a.out : $(OBJECT)
- gcc -o combine_wave $(OBJECT)
+ $(CC) $(LDFLAGS) -o combine_wave $(OBJECT)
# DEPENDENCIES
combine_wave.o : combine_wave.c combine_wave.h wave_header.h
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Re: [asterisk-users] Asterisk 1.4 Call Recording
Never mind, I got it. I needed a -p0
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 15, 2008 11:19 AM
Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording
Hi Mike,
On Tue, 2008-01-15 at 10:57 -0600, Mike Hammett wrote:
Are there any tricks to getting combine_wave to make?
Patch attached. Builds fine with patch on Fedora 8.
Regards,
Patrick
diff -Naur combine_wave-0.3.orig/combine_wave.c
combine_wave-0.3/combine_wave.c
--- combine_wave-0.3.orig/combine_wave.c 2005-10-06 14:44:10.0
+0200
+++ combine_wave-0.3/combine_wave.c 2007-10-05 21:02:17.0 +0200
@@ -19,8 +19,8 @@
void running_info()
{
-fprintf(stderr,\
-RUNNNING COMMANDS
+fprintf(stderr,
+RUNNNING COMMANDS\n\
b toggles move both channels / move right channel delay mode.\n\
ESC exits.\n\
'z' 'x' 1 sample forward / backward.\n\
@@ -39,8 +39,8 @@
void usage()
{
-fprintf(stderr,\
-Usage:
+fprintf(stderr,
+Usage:\n\
combine_wave [-a] [-d milli seconds delay right channel relative to
left]\n\
[-e samples delay right channel relative to left]\n\
[-k] -l filename_left [-m] -o output_filename -r filename_right [s start
seek offset].\n\
diff -Naur combine_wave-0.3.orig/combine_wave.h
combine_wave-0.3/combine_wave.h
--- combine_wave-0.3.orig/combine_wave.h 2005-10-06 14:44:10.0
+0200
+++ combine_wave-0.3/combine_wave.h 2007-10-05 21:02:52.0 +0200
@@ -5,6 +5,7 @@
#include unistd.h
#include stdio.h
#include stdlib.h
+#include string.h
#include signal.h
#include errno.h
diff -Naur combine_wave-0.3.orig/Makefile combine_wave-0.3/Makefile
--- combine_wave-0.3.orig/Makefile 2005-10-06 14:44:10.0 +0200
+++ combine_wave-0.3/Makefile 2007-10-05 21:00:43.0 +0200
@@ -6,13 +6,13 @@
CFLAGS = -O2 -Wall -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
.c.o:
- gcc $(CFLAGS) -c $
+ $(CC) $(CFLAGS) -c $
OBJECT =\
combine_wave.o
a.out : $(OBJECT)
- gcc -o combine_wave $(OBJECT)
+ $(CC) $(LDFLAGS) -o combine_wave $(OBJECT)
# DEPENDENCIES
combine_wave.o : combine_wave.c combine_wave.h wave_header.h
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[asterisk-users] Asterisk 1.4 Call Recording
I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't
seem to get it to work correct. Could someone point me to what I need to do?
I have attached what I believe are the relevant parts.
[globals]
; script to be executed when monitoring has been finished
MONITOR_EXEC=/usr/local/bin/2wav2mp3
; uncomment this line if you are using Ogg Vorbis
;MONITOR_EXEC=/usr/local/bin/2wav2ogg
[test]
exten =
555,1,SetVar(CALLFILENAME=outgoing/${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP}-${EXTEN})
exten = 555,2,Monitor(wav,${CALLFILENAME},m)
exten = 555,3,Dial(IAX2/ics.iax-trunk/${EXTEN})
exten = 555,4,Hangup()
exten = 815787,1,Set(CALLFILENAME=outgoing/1815739)
exten = 815787,2,Monitor(wav,${CALLFILENAME},m)
exten = 815787,3,Dial(IAX2/ics.iax-trunk/1815739)
exten = 815787,4,Hangup()
[EMAIL PROTECTED] asterisk]# cat /usr/local/bin/2wav2mp3
#!/bin/sh
# 2wav2mp3 - create stereo mp3 out of two mono wav-files
# source files will be deleted
#
# 2005 05 23 dietmar zlabinger http://www.zlabinger.at/asterisk
# 2006 03 24 modified for sox 12.17.9 as of Suse9.2 by Matthias
#
# usage: 2wav2mp3 wave1 wave2 mp3
# designed for Asterisk Monitor(file,format,option) where option is e and
# the variable
# MONITOR_EXEC/usr/local/bin/2wav2mp3
# location of SOX and SOXMIX
# (set according to your system settings, eg. /usr/bin)
SOX=/usr/bin/sox
SOXMIX=/usr/bin/soxmix
# lame is only required when sox does not support liblame
LAME=/usr/local/bin/lame
# command line variables
LEFT=$1
RIGHT=$2
OUT=$3
LTMP=asename $1 .wavmp.wav
RTMP=asename $2 .wavmp.wav
#test if input files exist
test ! -r $LEFT exit
test ! -r $RIGHT exit
# convert mono to stereo, adjust balance to -1/1
# left channel
$SOX $LEFT -t wav -c 2 $LTMP pan -1
# right channel
$SOX $RIGHT -t wav -c 2 $RTMP pan 1
# combine and compress
# this requires sox to be built with mp3-support.
# To see if there is support for Mp3 run sox -h and
# look for it under the list of supported file formats as mp3.
#$SOXMIX -v 1 $LTMP -v 1 $RTMP -t mp3 -v 1 $OUT.mp3
# in case an old version of sox is used, encoding
# can be done afterwards
$SOXMIX -v 1 $LTMP -v 1 $RTMP -v 1 $OUT
$LAME -S -V7 -B24 --tt $OUT --add-id3v2 $OUT $OUT.mp3
#remove temporary files
test -w $LTMP rm $LTMP
test -w $RTMP rm $RTMP
test -w $OUT rm $OUT
#remove input files if successfull
test -r $OUT.mp3 rm $LEFT $RIGHT
# eof
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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Re: [asterisk-users] Asterisk 1.4 Call Recording
Does what I have in the dialplan look right or am I way off base with being able to use that script? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Steve Johnson [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 14, 2008 10:51 AM Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording You might take a few ideas from this combine.sh script which works for me. It uses the combine_wave program from http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame program to convert to mp3. It converts the entire directory /var/spool/asterisk/monitor/*-in.wav files to mp3 where the mp3 file doesn't already exist. S. File: combine.sh --- #!/bin/sh cd /var/spool/asterisk/monitor for f in *-in.wav do in=$f out=`echo $f | sed -e 's/-in.wav/-out.wav/'` tmpwav=`echo $f | sed -e 's/-in.wav/-both.wav/'` mp3=`echo $f | sed -e 's/-in.wav/.mp3/'` if [ -e $mp3 ] then continue fi # combine the two tracks into one stereo file /usr/local/bin/combine_wave -l $in -r $out -o $tmpwav 2/dev/null /usr/bin/lame --silent -h -b 96 $tmpwav $mp3 # Remove temporary .wav files test -w $tmpwav rm $tmpwav # Remove input files if successful test -s $mp3 rm $in $out done exit 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] e911
*bump* - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Tuesday, November 20, 2007 12:27 PM Subject: [asterisk-users] e911 One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] e911
Then I could just make downstream-phones my current outbound context and everything would do what I'm after. I got what you're saying. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Dave Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, November 24, 2007 2:25 PM Subject: Re: [asterisk-users] e911 Mike Hammett wrote on 11/20/07 1:27 PM: One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? I think the easiest answer is going to be to go ahead and put each in their own context. Note that you can include contexts from each other... so say they're all in [downstream-phones] right now (for example)... you can do something like this: [phones-in-account1] include = downstream-phones exten = 911,s,Goto(DialViaAccount1) [phones-in-account2] include = downstream-phones exten = 911,s,Goto(DialViaAccount2) etc. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] e911
One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different Networks
That server traceroutes out that interface. Yes, I can lynx to google.com. Braxis*CLI iax2 reload == Parsing '/etc/asterisk/iax.conf': Found doing dnsmgr_lookup for '208.100.1.33' doing dnsmgr_lookup for '208.100.1.33' == Parsing '/etc/asterisk/users.conf': Found == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00017ms SCall: 2 DCall: 0 [208.100.1.33:4569] USERNAME: rwestics REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00017ms SCall: 3 DCall: 0 [208.100.1.33:4569] USERNAME: ottos REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00017ms SCall: 9 DCall: 2 [208.100.1.33:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 9ms SCall: 9 DCall: 2 [208.100.1.33:4569] AUTHMETHODS : 3 CHALLENGE : 29638146 USERNAME: rwestics Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00035ms SCall: 2 DCall: 9 [208.100.1.33:4569] USERNAME: rwestics REFRESH : 60 MD5 RESULT : 1c113a5aaa20100f2c864544b892fea3 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00017ms SCall: 00013 DCall: 3 [208.100.1.33:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00010ms SCall: 00013 DCall: 3 [208.100.1.33:4569] AUTHMETHODS : 3 CHALLENGE : 43340858 USERNAME: ottos Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 00036ms SCall: 3 DCall: 00013 [208.100.1.33:4569] USERNAME: ottos REFRESH : 60 MD5 RESULT : 4d18ad3b06bc96496f59655367093ecf Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00035ms SCall: 9 DCall: 2 [208.100.1.33:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00036ms SCall: 9 DCall: 2 [208.100.1.33:4569] USERNAME: rwestics DATE TIME : 2007-09-13 07:00:54 REFRESH : 60 APPARENT ADDRES : IPV4 24.14.116.22:4569 CALLING NUMBER : 8159092441 CALLING NAME: West and Associates -- Registered IAX2 to '208.100.1.33', who sees us as 24.14.116.22:4569 with no messages waiting Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00036ms SCall: 2 DCall: 9 [208.100.1.33:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00036ms SCall: 00013 DCall: 3 [208.100.1.33:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00040ms SCall: 00013 DCall: 3 [208.100.1.33:4569] USERNAME: ottos DATE TIME : 2007-09-13 07:00:54 REFRESH : 60 APPARENT ADDRES : IPV4 24.14.116.22:4569 CALLING NUMBER : 8157582715 CALLING NAME: Ottos Nightclub -- Registered IAX2 to '208.100.1.33', who sees us as 24.14.116.22:4569 with no messages waiting Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00040ms SCall: 3 DCall: 00013 [208.100.1.33:4569] Braxis*CLI iax2 no debug IAX2 Debugging Disabled The 'iax2 no debug' command is deprecated and will be removed in a future release. Please use 'iax2 set debug off' instead. Braxis*CLI iax2 show peers Name/UsernameHost Mask Port Status rwestics (Unspecified) (D) 255.255.255.255 0 Unmonitored rwest1/rwest1(Unspecified) (D) 255.255.255.255 0 Unmonitored 224/224 (Unspecified) (D) 255.255.255.255 0 Unmonitored ics/ottos(Unspecified) (D) 255.255.255.255 0 UNKNOWN 4 iax2 peers [0 online, 1 offline, 3 unmonitored] That's what happens after I do a iax2 show peers. So apparently calls are coming in, but showing the peers isn't bringing up any IP addresses. I can also make outbound calls. So... apparently Asterisk is working except for the servers aren't showing up in the peer list. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Erik Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 12, 2007 12:44 PM Subject: Re: [asterisk-users] Different Networks On 9/7/07, Mike Hammett [EMAIL PROTECTED] wrote: If it has nothing to do with Asterisk, then why does every other device work as its supposed to? You never answered
Re: [asterisk-users] Different Networks
*bump* - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 07, 2007 3:25 PM Subject: Re: [asterisk-users] Different Networks If it has nothing to do with Asterisk, then why does every other device work as its supposed to? An MGCP ATA routes out that interface. A laptop routes out that interface. That server traceroutes out that interface. Asterisk doesn't link up. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Erik Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 07, 2007 3:06 PM Subject: Re: [asterisk-users] Different Networks On 9/6/07, Mike Hammett [EMAIL PROTECTED] wrote: I have multiple upstreams in my office. The primary upstream is having some issues with latency\jitter. I want to move the VoIP traffic to another interface. I have the router set to send all traffic destined for local networks out the respective interfaces. Traffic destined to the Internet goes out one of the upstreams. I can do this on a per-IP basis and have successfully done so in testing on my laptop and a couple other machines. I also have it in production for an ATA. I also switch all devices to use another upstream with the failure of the primary ISP. Again, this works with everything but the Asterisk server. The internal Asterisk server cannot connect to the Asterisk server out on the public Internet. How do I investigate this? Mike - there's no reason this routing problem would have anything to do with asterisk itself.Have you tried running links (or another text web browser) on the asterisk server to see if you're able to get traffic past the gateway? Do you have the default gateway and/or routing tables configured correctly on the asterisk server? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different Networks
*bump* - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Thursday, September 06, 2007 10:05 AM Subject: [asterisk-users] Different Networks I have multiple upstreams in my office. The primary upstream is having some issues with latency\jitter. I want to move the VoIP traffic to another interface. I have the router set to send all traffic destined for local networks out the respective interfaces. Traffic destined to the Internet goes out one of the upstreams. I can do this on a per-IP basis and have successfully done so in testing on my laptop and a couple other machines. I also have it in production for an ATA. I also switch all devices to use another upstream with the failure of the primary ISP. Again, this works with everything but the Asterisk server. The internal Asterisk server cannot connect to the Asterisk server out on the public Internet. How do I investigate this? Here is the definition on the internal server: [rwestics] type=friend ;host=208.100.1.33 ;miho.ics-il.net host=dynamic ;username=rwestics secret=*** context=rwest disallow=all allow=ulaw Here is the definition on the public Internet server: [rwestics] type=friend host=dynamic ;username=ics secret=** qualify=yes context=outbound-scripted accountcode=12 callerid=* - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different Networks
If it has nothing to do with Asterisk, then why does every other device work as its supposed to? An MGCP ATA routes out that interface. A laptop routes out that interface. That server traceroutes out that interface. Asterisk doesn't link up. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Erik Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 07, 2007 3:06 PM Subject: Re: [asterisk-users] Different Networks On 9/6/07, Mike Hammett [EMAIL PROTECTED] wrote: I have multiple upstreams in my office. The primary upstream is having some issues with latency\jitter. I want to move the VoIP traffic to another interface. I have the router set to send all traffic destined for local networks out the respective interfaces. Traffic destined to the Internet goes out one of the upstreams. I can do this on a per-IP basis and have successfully done so in testing on my laptop and a couple other machines. I also have it in production for an ATA. I also switch all devices to use another upstream with the failure of the primary ISP. Again, this works with everything but the Asterisk server. The internal Asterisk server cannot connect to the Asterisk server out on the public Internet. How do I investigate this? Mike - there's no reason this routing problem would have anything to do with asterisk itself.Have you tried running links (or another text web browser) on the asterisk server to see if you're able to get traffic past the gateway? Do you have the default gateway and/or routing tables configured correctly on the asterisk server? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different Networks
I have multiple upstreams in my office. The primary upstream is having some issues with latency\jitter. I want to move the VoIP traffic to another interface. I have the router set to send all traffic destined for local networks out the respective interfaces. Traffic destined to the Internet goes out one of the upstreams. I can do this on a per-IP basis and have successfully done so in testing on my laptop and a couple other machines. I also have it in production for an ATA. I also switch all devices to use another upstream with the failure of the primary ISP. Again, this works with everything but the Asterisk server. The internal Asterisk server cannot connect to the Asterisk server out on the public Internet. How do I investigate this? Here is the definition on the internal server: [rwestics] type=friend ;host=208.100.1.33 ;miho.ics-il.net host=dynamic ;username=rwestics secret=*** context=rwest disallow=all allow=ulaw Here is the definition on the public Internet server: [rwestics] type=friend host=dynamic ;username=ics secret=** qualify=yes context=outbound-scripted accountcode=12 callerid=* - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ping
- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
I've been trying to send messages to the list for the past 24 hours, but they just aren't going through. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 7:23 AM Subject: [asterisk-users] Ping - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
Agreed. This conversation is working just fine, but the important messages I'm trying to get to go through aren't. I've never had consistent success from posting to asterisk-users. Asterisk-biz seems to work all of the time. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Sander Smeenk [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 7:45 AM Subject: Re: [asterisk-users] Ping Quoting Doug Lytle ([EMAIL PROTECTED]): Pong The list seems to act weird. I mailed to the list earlier, the message was accepted, but does not appear on the archives nor did i get a bounce or my own listmail back. Though i do see other people posting :/ -- | Only those who will risk going too far, can possibly find out how far you can go. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
*nods* I verified more than once and even copied + pasted to make sure. Obviously my ping message went through, but my others have not. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Bill Andersen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 9:04 AM Subject: Re: [asterisk-users] Ping Dave Fullerton wrote: Same thing happened to me a while back. I sent a new message asking a question ..twice.. and neither made it through. However replies to other peoples messages went through just fine. This may not be the problem, but I've seen this on my NEW post a few times and it was always my fault. My default email is NOT the email I have subscribed to this list. Only subscribers can post. Others don't seem to bounce (why bounce to a spammer) and they are just dropped. However, when I reply to a post, it uses the correct address automatically because the original email originated from the list (with my subscribed address). Make sure your NEW posts are sent from the subscribed address... Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
and I appreciate it much. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Jared Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 05, 2007 9:25 AM Subject: Re: [asterisk-users] Ping On Wed, 2007-09-05 at 09:11 -0500, Mike Hammett wrote: *nods* I verified more than once and even copied + pasted to make sure. Obviously my ping message went through, but my others have not. I'm working with Digium's IT department to try to track down the problem. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi port IAX Gateway
I am looking for a gateway that has several FXS ports and uses IAX. I have a need for 16 ports, but will accept 6 or 8 port gateways as well. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 320 messages
I used this site (and perhaps a couple other Google returned) as well as the Polycom Admin guide as reference. http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+wit h+Asterisk 299 is the extension one dials to access their voicemail (with caller ID sending to the correct voicemail). I see the bold on the wiki page telling me to put in my voicemail context, but I'm not sure where they're talking about. Previous to doing this work, the phone said there were two voicemails when there were none. Now it doesn't say there are voicemails when they are there. This is the entry in the sip.conf [rwest200] type=friend secret=abc123 context=rwest host=dynamic [EMAIL PROTECTED] callerid=Rob West 200 username=rwest200 qualify=no port=5060 nat=no dtmfmode=rfc2833 canreinvite=no This is the voicemail.conf [rwest] 200 = 1234,Rob West 201 = 1234,Julia Zeiter 202 = 1234,Larry Sallberg This is the phonex.cfg ?xml version=1.0 encoding=UTF-8 standalone=yes? !-- Per-phone configuration in this file -- reginfo reg reg.1.displayName=200 Rob West reg.1.address=rwest200 reg.1.label=200 reg.1.auth.userId=rwest200 reg.1.auth.password=abc123/ /reginfo msg msg.bypassInstantMessage=1 msg.mwi.1.subscribe= msg.mwi.1.callBack=299 msg.mwi.1.callBackMode=contact/ /msg - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Verizon Interconnection
Now that MCI and Verizon are one, they're probably on legacy MCI. MCI was also the one that was doing the wholesale SIP pre-merger. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, June 06, 2007 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Verizon Interconnection Verizon has phone service in Switzerland? Or are you getting US numbers? On 6/6/07, laurent schweizer [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: HI, yes we are interconnected with Verizon in SIP, but we are in europe (Switzerland) so I don't know If it is the same process in USA ... Laurent 2007/6/6, Matt [EMAIL PROTECTED]: So absolutely no one here was interconnected with Verizon? I am going to shoot this over to asterisk-biz, also, in hopes someone may have missed it that is on the biz list. The question again is: Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Oddity
Why would calls be coming in on the Guest IAX account? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Monday, June 04, 2007 6:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Oddity I have two Asterisk servers. One is my primary server that I link to all of my providers and the other is at an office building with multiple tenants. If I tell Asterisk to dial an entry in the iax.conf that is for one customer off that second box, why does it use a different account for a different customer? It still ends up at the correct box, but it is hard to troubleshoot issues when calls are coming in under different accounts (probably different account codes too). Also, the second customer on that box. Earlier today everything worked fine as was. Later all calls going to that customer were going to the default context, despite the fact that I explicitly defined the context I wanted the calls to go to in all entries in iax.conf. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Oddity
I have two Asterisk servers. One is my primary server that I link to all of my providers and the other is at an office building with multiple tenants. If I tell Asterisk to dial an entry in the iax.conf that is for one customer off that second box, why does it use a different account for a different customer? It still ends up at the correct box, but it is hard to troubleshoot issues when calls are coming in under different accounts (probably different account codes too). Also, the second customer on that box. Earlier today everything worked fine as was. Later all calls going to that customer were going to the default context, despite the fact that I explicitly defined the context I wanted the calls to go to in all entries in iax.conf. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HP OfficeJet 6110, Sipura 2102, T.38, and Clarent
I have a Sipura 2102 that I'm trying to do T.38 with a Clarent C5CM. I figured skipping the Asterisk server and going right to the Clarent would be best as T.38 is new in Asterisk. However, I can't get the Sipura to link up with the Clarent. Sipura support is less than responsive. Note: I set myself up as a Linksys Partner, and have spent hour(s) on the phone with them, but it still doesn't work. Ideas? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Caller ID matching
Yeah, I was trying to have it match the caller ID with what they're dialing
so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham
Sent: Tuesday, May 22, 2007 5:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID matching
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, Mike Hammett mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number to
check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected connect
attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]' does not
exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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[asterisk-users] VoiceMail Access
I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at ATT, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when you hear your greeting when calling yourself Toll free number What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoiceMail Access
If it is easy, could you enlighten me? I have another thread on caller ID
matching, but I haven't received any positive responses.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent: Monday, May 21, 2007 5:55 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] VoiceMail Access
On Mon, May 21, 2007 at 03:10:23PM -0400, Lee Jenkins wrote:
Mike Hammett wrote:
I was looking at the ILECs' web sites to determine how their users
access voicemail.
What method should I use for my users checking their voicemail? Can
Asterisk voicemail be made to accept hitting * during the greeting to
enter the voicemail system? If they call their own number, how do I get
Asterisk to recognize that and take them to the voicemail system?
A common approach is to use the caller id in combination with some digit
sequence. For my systems, I've just used 555 as the VM extension.
exten=555,1,VoicemailMain(${CALLERID(num)})
For access to the VM from outside the system, I've used an AGI script to
query a database to validate the user.
It's also quite easy to set-up if you call your own extension number
from your extension it goes into voicemail for you extension.
You can have another number as above to access voicemail from another
extension.
Steve
--
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UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
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[asterisk-users] Caller ID matching
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number to
check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected connect
attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]' does not
exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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RE: [asterisk-users] SIP NAT
If I have several local networks, can I specify that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, March 29, 2007 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP NAT Mike Hammett wrote: I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server - MT w\NAT - PPPoE - MT - Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk -- PSTN works great. Audio Asterisk -- PSTN does not. That would be expected since you did not forward the ports used for RTP. See /etc/asterisk/rtp.conf A sample is in the Asterisk source. Did you also set localnet= and externip= options in sip.conf [general]. SIP works just fine with NAT if you have it correctly configured and your server is on a static IP address. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP NAT
I checked into it and it seems to recognize multiple entries as debug displays it. --Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Friday, March 30, 2007 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP NAT According to sip.conf.sample the answer is...well, I guess you can look in /path/to/src/asterisk/configs/sip.conf.sample and see for yourself. Mike Hammett wrote: If I have several local networks, can I specify that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, March 29, 2007 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP NAT Mike Hammett wrote: I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server - MT w\NAT - PPPoE - MT - Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk -- PSTN works great. Audio Asterisk -- PSTN does not. That would be expected since you did not forward the ports used for RTP. See /etc/asterisk/rtp.conf A sample is in the Asterisk source. Did you also set localnet= and externip= options in sip.conf [general]. SIP works just fine with NAT if you have it correctly configured and your server is on a static IP address. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP NAT
I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server - MT w\NAT - PPPoE - MT - Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk -- PSTN works great. Audio Asterisk -- PSTN does not. Ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP NAT
I have a block of 20 IP addresses, so I can't really carve out /30s and whatnot to route between locations. Asterisk is the client. I am doing Interop testing with some vendors before I ship it out to a colo facility. I have used the NAT setting with Asterisk as the server on the open Internet. Would it function similarly with Asterisk as the client? We are using IP based authentication. I have more public addresses, but I'm unsure how to route that through so the Asterisk box can use it. Perhaps I will look into 1:1 NAT. --Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Thursday, March 29, 2007 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SIP NAT What do you mean by 'non-standard' IP block? Is the Asterisk machine behind a NAT, or are only your clients? Did you look at the nat setting sin sip.conf? Do you have a static public address that can be routed to the Asterisk box? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Thursday, March 29, 2007 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] SIP NAT I hate SIP. The only reason I'm doing this is that its cheaper than deploying the server to a colo facility. My provider has given me a non-standard IP block, so I can't do typical routing. I have Asterisk server - MT w\NAT - PPPoE - MT - Provider. I setup a dst-nat on 5060 to the Asterisk box. Audio from Asterisk -- PSTN works great. Audio Asterisk -- PSTN does not. Ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] help - UNSUBSCRIBE
Good job on reading the line at the top of the digest on how to unsubscribe. --Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerric Sent: Thursday, March 29, 2007 11:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] help - UNSUBSCRIBE Please remove this email from your mailing list. UNSUBSCRIBE Thank you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 29, 2007 9:14 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 32, Issue 118 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Re: Re: Inbound Voice Quality - Speed Change (Tzafrir Cohen) 2. Re: error in FreePBX (Steve Murphy) 3. SV: [asterisk-users] Set(CALLERID(all) not working with 'unknown'call? ([EMAIL PROTECTED]) 4. Re: Transfering not working - how to debug? (Rizwan Hisham) 5. Off Topic: Open Source USB Softphone (Luis Claudio Santos) 6. Where are Spandsp changelogs or bugs available ? (Olivier) 7. L options in Dial() dont seem to work (Mark Reardon) 8. maximum simultaneous calls (Mark Quitoriano) 9. Re: L options in Dial() dont seem to work (Eric ManxPower Wieling) 10. Asterisk does not reINVITE after 302Redirect 401Unauthorized ([EMAIL PROTECTED]) 11. Re: L options in Dial() dont seem to work (Steve Murphy) 12. Is it possible to install CCM on a Linux platform ? (Olivier) 13. Re: L options in Dial() dont seem to work (Mark Reardon) 14. Scratchy Audio with Asterisk 1.2.4 over IAX onFreeBSD? (Benoit Panizzon) 15. Re: Cisco 30VIP Phone (Jason Parker) 16. SIP NAT (Mike Hammett) 17. Re: maximum simultaneous calls (Matthew J. Roth) 18. RE: SIP NAT (Alexander Lopez) 19. Re: Multi-line phones - Asterisk uses wrong callerid (Drew Gibson) -- Message: 1 Date: Thu, 29 Mar 2007 15:40:20 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Re: Re: Inbound Voice Quality - Speed Change To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Thu, Mar 29, 2007 at 08:28:53AM -0400, Jim Duda wrote: The zttest program results in 99%. So you have a working timing source. No need to waste your time here. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- Message: 2 Date: Thu, 29 Mar 2007 07:59:34 -0600 From: Steve Murphy [EMAIL PROTECTED] Subject: Re: [asterisk-users] error in FreePBX To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8 On Thu, 2007-03-29 at 13:26 +0200, Remco Barendse wrote: On Thu, 29 Mar 2007, Carlos JerC3nimo wrote: Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg Im-sorryan-error-has-occured and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some NAT configuration parameters, it didnt worked aswell. As this ever happened to anyone before? Any hints are very appreciated. Thank you very much I have the same problem, it seems to occur when an extension is busy here. All my extensions are on local lan with phones having ip addresses in a private range without NAT or anything so that is not the problem. Sounds like an error in the dial pan FreePBX generated. My suggestion: try a FreePBX mailing list first; the problem *is* more likely to be in their stuff. murf -- Steve Murphy Software Developer Digium -- next part -- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3227 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/61f67f 5c/smime-0001.bin -- Message: 3 Date: Thu, 29 Mar 2007 16:04:43 +0200 From: [EMAIL PROTECTED] Subject: SV: [asterisk-users] Set(CALLERID(all) not working with 'unknown'call? To: asterisk-users
[asterisk-users] Polycom Power
I have a 501 with traditional power and a 301 with PoE. I rightfully assumed that the traditional power from the 501 would work on the 301. How do I get the PoE to work? Do I use the Polycom PoE cable in addition to whatever PoE injection method I use? I have a Cisco PoE injector that works on my Cisco AP350 and my 7960. No combination of this injector, the Polycom cable, and the phone result in success. I have 18v PoE injectors that I use for other things, but I hear that 802.3af is 48v, therefore probably wouldn't work. How do I use Polycom PoE? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom and Asterisk
I was previously having an issue with a Polycom phone and Polycom support said that Asterisk didn't play well with Polycom firmware versions 1.6.7 and newer due to SIP compatibility issues. I believe I heard a lot of things were fixed\adjusted in 1.4 and was wondering if anyone has had success with Asterisk 1.4 and the latest Polycom firmware releases. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom and Asterisk
To be honest, I don't remember. It may have been mentioned in passing by the tech. However, one of my clients with Polycom phones was having a problem that I cannot now recall and going back to 1.6.7 fixed it. I'll try again with 2.1. --Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Wednesday, March 28, 2007 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and Asterisk Matt, I am running Polycom 2.1 on both 1.4 and 1.2 svn releases without any problems. What kind of issues did you experience? On 3/28/07, Mike Hammett [EMAIL PROTECTED] wrote: I was previously having an issue with a Polycom phone and Polycom support said that Asterisk didn't play well with Polycom firmware versions 1.6.7 and newer due to SIP compatibility issues. I believe I heard a lot of things were fixed\adjusted in 1.4 and was wondering if anyone has had success with Asterisk 1.4 and the latest Polycom firmware releases. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Realtime
I enabled some more detailed debugging and logging as per someone else a few posts ago and I saw that the permissions on MySQL were set incorrectly. I granted all, but what are the least permissions this user should need? How do I register to other servers? It seems to be ignoring the register statements in my iax.conf. --Mike All that looks fine. What do you get when you do realtime mysql status? The next areas to look at would be your DB configs, and debug status when you actually try to use one of the entries in your DB. . . I only use it for iaxpeers/users and extensions, so I can't comment much on its use with SIP or voicemail. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime
[EMAIL PROTECTED] asterisk]# cat res_mysql.conf ; ; Sample configuration for res_config_mysql.c ; ; The value of dbhost may be either a hostname or an IP address. ; If dbhost is commented out or the string localhost, a connection ; to the local host is assumed and dbsock is used instead of TCP/IP ; to connect to the server. ; [general] ;dbhost = 127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = yourpassword ;dbport = 3306 dbsock = /var/lib/mysql/mysql.sock [EMAIL PROTECTED] asterisk]# cat extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/extconfig.txt for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using 'preload' statements in modules.conf: ; manager.conf ; cdr.conf ; rtp.conf ; ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;example = odbc,asterisk,alttable ;example2 = ldap,dc=oxymium,dc=net,example2 ; ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk ;sipusers = odbc,asterisk ;sippeers = odbc,asterisk ;voicemail = odbc,asterisk ;extensions = odbc,asterisk ;queues = odbc,asterisk ;queue_members = odbc,asterisk sippeers = mysql,asterisk,sip_peers sipusers = mysql,asterisk,sip_users iaxpeers = mysql,asterisk,iax_peers iaxusers = mysql,asterisk,iax_users queues = mysql,asterisk,queue_table queue_members = mysql,asterisk,queue_member_table voicemail = mysql,asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Realtime
== Parsing '/etc/asterisk/res_mysql.conf': [Mar 7 14:12:37] DEBUG[4380]: config.c:844 config_text_file_load: Parsing /etc/asterisk/res_mysql.conf Found [Mar 7 14:12:37] WARNING[4380]: res_config_mysql.c:555 parse_config: MySQL RealTime: No database host found, using localhost via socket. [Mar 7 14:12:37] WARNING[4380]: res_config_mysql.c:555 parse_config: MySQL RealTime: No database host found, using localhost via socket. [Mar 7 14:12:37] WARNING[4380]: res_config_mysql.c:569 parse_config: MySQL RealTime: No database port found, using 3306 as default. [Mar 7 14:12:37] WARNING[4380]: res_config_mysql.c:569 parse_config: MySQL RealTime: No database port found, using 3306 as default. [Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:585 parse_config: MySQL RealTime Host: [Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:586 parse_config: MySQL RealTime Port: 3306 [Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:590 parse_config: MySQL RealTime User: asterisk [Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:591 parse_config: MySQL RealTime Password: yourpassword [Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:623 mysql_reconnect: MySQL RealTime: Successfully connected to database. [Mar 7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register: Registered Config Engine mysql [Mar 7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register: Registered Config Engine mysql MySQL RealTime driver loaded. res_config_mysql.so = (MySQL RealTime Configuration Driver) -- Message: 5 Date: Thu, 01 Mar 2007 20:58:37 +0100 From: Philipp Kempgen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Realtime To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-15 Brian Capouch wrote: Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. You don't say which version you're running. I *think* the syntax is the same for both: realtime driver-name status will show you the status. For postgres it's pgsql for driver name (that's what I use). I think the other driver ids are mysql and odbc. If you don't see yourself connected, that's where to start. Or put console = notice,warning,error,verbose,debug in logger.conf / run asterisk -vvvdddc :) This will give you all MySQL queries and warnings. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschdftsf|hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP addresses
I have multiple IP addresses on my box. My provider just changed my eth0 IP off to another interface (lo:9) and a new IP on eth0. Nothing works anymore because calls to the old IP address are being answered by the new IP address. How do I straighten this out? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.0
I believe I noticed that I had upgraded the kernel, but not yet restarted. I restarted, and I think that was all I had to do to get it running again. --Mike Message: 14 Date: Wed, 21 Feb 2007 18:01:22 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] Zaptel 1.4.0 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Wed, Feb 21, 2007 at 09:52:32AM -0600, Mike Hammett wrote: I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make install and I don't see any errors. This is out of my modprobe.conf: [ snip ] However: [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel FATAL: Module zaptel not found. Any chance that this is just a missing depmod run? depmod modinfo zaptel Or maybe you installed the modules to an incorrect directory: uname -r find /lib/modules -name zaptel.ko If so, it probably means you built it with incorrect kernel source / configuration. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Snom 320 password
It must be the challenge response bug. They are still using 1.0.x and I turned off challenge response on the phone. I made the change last week, but I haven't heard from the user one way or the other. This server is slated to be upgraded to 1.4.0. --Mike -- Message: 14 Date: Wed, 21 Feb 2007 16:52:10 -0600 From: Mike Hammett [EMAIL PROTECTED] Subject: [asterisk-users] Snom 320 password To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii A client of mine has a Snom 320. Usually when he comes in each morning, it is asking him for a password. A power cycle brings it back to normal operation. How do I troubleshoot this further? --Mike -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070221/0e4d3a b8/attachment-0001.htm -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime
Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime
queue show Show status of a specified queue realtime load Used to print out RealTime variables. realtime update Used to update RealTime variables. restart gracefully Restart Asterisk gracefully Aiur*CLI realtime load You must supply a family name, a column to match on, and a value to match to. I am using Asterisk 1.4.0 and MySQL. It appears that the only realtime options are for loading and updating specific items from the database. The only database options seem to be for dundi. Under modules, all I could find is: Aiur*CLI module show like pbx_realtime.so Module Description Use Count pbx_realtime.soRealtime Switch 0 1 modules loaded --Mike -- Message: 12 Date: Thu, 01 Mar 2007 13:02:23 -0500 From: Brian Capouch [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Realtime To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252; format=flowed Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I cant see any signs that its working. I followed and double-checked a few different guides around the net, but havent been able to figure it out. You don't say which version you're running. I *think* the syntax is the same for both: realtime driver-name status will show you the status. For postgres it's pgsql for driver name (that's what I use). I think the other driver ids are mysql and odbc. If you don't see yourself connected, that's where to start. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AG-188
I do believe it is that chipset. The person placing the call from the AG-188 does not hear a ring. --Mike Message: 8 Date: Fri, 23 Feb 2007 01:21:52 + From: Thomas Kenyon [EMAIL PROTECTED] Subject: Re: [asterisk-users] AG-188 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252; format=flowed Mike Hammett wrote: Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (havent tried SIP), there is no ring. Is this that you hear no ring or the other end doesn't ring? From vague memory the AG-188 is an Infineon chipset ATA (which I haven't used.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AG-188
Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (haven't tried SIP), there is no ring. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk CDR MySQL
I removed Asterisk and reinstalled it from scratch. It seems to be working now as module show like cdr now reports many more lines and now mentions MySQL. The database is the same as I didn't remove that, just the various files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, February 21, 2007 3:12 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 31, Issue 90 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Mask the caller-ID (Joanna Liza Mariazeta) 2. The High Performance Echo Canceller (HPEC) (Boris Bakchiev) 3. Asterisk behind OpenSER - Getting SIP reinvites towork with an ITSP (Hugo Livude) 4. Re: Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 (Tzafrir Cohen) 5. Re: They ignore my DTMF! (Pierre Marceau) 6. Re: Passing a variable from one Asterisk box to another (Justin Newman) 7. Re: They ignore my DTMF! (Benjamin Jacob) 8. Re: Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 (Eric ManxPower Wieling) 9. Re: They ignore my DTMF! (Pierre Marceau) 10. Re: Best FXO Gateway (Martin Joseph) 11. Re: They ignore my DTMF! (Benjamin Jacob) 12. Help! How to get ANSWEREDTIME after DIAL a ZAPchannel? (Charles Wang) 13. Re: Asterisk CDR MySQL (Goke Aruna) 14. Open Source VOIP at Toronto Conference (Evan Leibovitch) 15. How to repeat pri show span and zap show channel commands (Olivier) 16. Re: They ignore my DTMF! (Joanna Liza Mariazeta) 17. How to read pri intense debug span data ? (Olivier) 18. Re: How to repeat pri show span and zap show channel commands (Tzafrir Cohen) 19. Re: Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 (Tzafrir Cohen) 20. Hint a sip account (Christian Gansberger) 21. How to read channel occupation from PRI INTENSE DEBUG ? (Olivier) -- Message: 13 Date: Wed, 21 Feb 2007 07:48:46 +0100 From: Goke Aruna [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk CDR MySQL To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252 Mike Hammett wrote: Im attempting to setup Asterisk 1.4.0 CDRs to use MySQL. Modules show like cdr_mysql.so tells me it is loaded. Reload cdr with MySQL started or stopped makes no difference in the errors. Ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users do you have cdr_mysql.conf well configured and write permmission granted to sql user.? give a verbose and debug to ur logger to know whether asterisk is attempting login or not. goksie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.4.0
I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make install and I don't see any errors. This is out of my modprobe.conf: install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp /sbin/ztcfg install pciradio /sbin/modprobe --ignore-install pciradio /sbin/ztcfg install ztd-loc /sbin/modprobe --ignore-install ztd-loc /sbin/ztcfg install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg alias wcfxs wctdm alias wct2xxp wct4xxp install zttranscode /sbin/modprobe --ignore-install zttranscode /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp /sbin/ztcfg However: [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel FATAL: Module zaptel not found. /var/log/dmesg doesn't say anything about zaptel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 320 password
A client of mine has a Snom 320. Usually when he comes in each morning, it is asking him for a password. A power cycle brings it back to normal operation. How do I troubleshoot this further? --Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CDR MySQL
I'm attempting to setup Asterisk 1.4.0 CDRs to use MySQL. Modules show like cdr_mysql.so tells me it is loaded. Reload cdr with MySQL started or stopped makes no difference in the errors. Ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNOW Migration
I currently have a customer that a previous employee setup with Gentoo\Asterisk. I'm looking to migrate to AsteriskNOW. They have a custom menu, which I would assume is easily replicable in AsteriskNOW. The only other thing I can think of is the sound bites for the menus. Does anyone have any advise or migration recommendations for this move? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Snom 320 echo
Where do I find more out in regards to the echo-cancelling component you mentioned? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 23, 2007 8:08 PM Subject: asterisk-users Digest, Vol 30, Issue 95 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. [OT] Mark Spencer Presents AsteriskNOW on Youtube (Damian Fossi) 2. RE: TDM2400 Hardware Echo Cancel (Webster, Andrew) 3. Re: How to exit from console? (Tzafrir Cohen) 4. OT: High Quality Wireless Headset for Cisco IP Phones and * (Tom) 5. RE: OT: High Quality Wireless Headset for Cisco IPPhones and * (Cory Andrews) 6. Re: Re: Dial plan constructions suggestions? (Lacy Moore - Aspendora) 7. Re: weird undocumented extensions such as s-BUSY (Barzilai Spinak) 8. Re: TDM2400 Hardware Echo Cancel (Mailing List) 9. RE: OT: High Quality Wireless Headset for Cisco IPPhones and * (Colin Anderson) 10. Re: Echo... (Matthew Fredrickson) 11. Snom 320 echo (Mike Hammett) 12. RE: Snom 320 echo (Colin Anderson) 13. AW: [asterisk-users] Snom 320 echo (Christian Stredicke) 14. automon and MONITOR_EXEC (John Williams) 15. DB_DELETE Function in 1.4 (Jeremiah Millay) 16. RE: * 1.0.9 Voicemail record name does not playb ack in Directory() --solved (Colin Anderson) 17. Re: How to exit from console? (Paul Hales) 18. Problem connecting PAP2 over wifi bridge (Alfredo Manrique) 19. DeStar 0.2.2 released! (Santiago Jos? Ruano Rinc?n) 20. Re: [OT] Mark Spencer Presents AsteriskNOW on Youtube (Dovid B) 21. Echo on IP phones... (Carlos Chavez) 22. Re: [OT] Mark Spencer Presents AsteriskNOW on Youtube (Kristian Kielhofner) 23. Re: Re: [asterisk-users] How to exit from console? ( ?? ) 24. Re: DB_DELETE Function in 1.4 (Alvin Austin) 25. Re: How to exit from console? (John Novack) 26. Re: realtime sipusers and rtcachefriends... bigheadache!! (kjcsb) 27. cmd Backgound problem with option m (Franz Wu) -- Message: 11 Date: Tue, 23 Jan 2007 15:10:28 -0600 From: Mike Hammett [EMAIL PROTECTED] Subject: [asterisk-users] Snom 320 echo To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070123/ef7ed599/attachment-0001.htm -- Message: 12 Date: Tue, 23 Jan 2007 14:21:52 -0700 From: Colin Anderson [EMAIL PROTECTED] Subject: RE: [asterisk-users] Snom 320 echo To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Later firmware versions have an echo-cancelling component in it, upgrade to latest version and also turn down the gains on the mic, the default setting is way too high. A setting of 3 or 4 max is all that is nessisary. hth -Original Message- From: Mike Hammett [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 23, 2007 2:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Snom 320 echo Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070123/22d41817/attachment-0001.htm -- Message: 13 Date: Tue, 23 Jan 2007 22:30:44 +0100 (MET) From: Christian Stredicke [EMAIL PROTECTED] Subject: AW: [asterisk-users] Snom 320 echo To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
[asterisk-users] RE: Snom 320 echo
Where do I find more out in regards to the echo-cancelling component you mentioned? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 23, 2007 8:08 PM Subject: asterisk-users Digest, Vol 30, Issue 95 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. [OT] Mark Spencer Presents AsteriskNOW on Youtube (Damian Fossi) 2. RE: TDM2400 Hardware Echo Cancel (Webster, Andrew) 3. Re: How to exit from console? (Tzafrir Cohen) 4. OT: High Quality Wireless Headset for Cisco IP Phones and * (Tom) 5. RE: OT: High Quality Wireless Headset for Cisco IPPhones and * (Cory Andrews) 6. Re: Re: Dial plan constructions suggestions? (Lacy Moore - Aspendora) 7. Re: weird undocumented extensions such as s-BUSY (Barzilai Spinak) 8. Re: TDM2400 Hardware Echo Cancel (Mailing List) 9. RE: OT: High Quality Wireless Headset for Cisco IPPhones and * (Colin Anderson) 10. Re: Echo... (Matthew Fredrickson) 11. Snom 320 echo (Mike Hammett) 12. RE: Snom 320 echo (Colin Anderson) 13. AW: [asterisk-users] Snom 320 echo (Christian Stredicke) 14. automon and MONITOR_EXEC (John Williams) 15. DB_DELETE Function in 1.4 (Jeremiah Millay) 16. RE: * 1.0.9 Voicemail record name does not playb ack in Directory() --solved (Colin Anderson) 17. Re: How to exit from console? (Paul Hales) 18. Problem connecting PAP2 over wifi bridge (Alfredo Manrique) 19. DeStar 0.2.2 released! (Santiago Jos? Ruano Rinc?n) 20. Re: [OT] Mark Spencer Presents AsteriskNOW on Youtube (Dovid B) 21. Echo on IP phones... (Carlos Chavez) 22. Re: [OT] Mark Spencer Presents AsteriskNOW on Youtube (Kristian Kielhofner) 23. Re: Re: [asterisk-users] How to exit from console? ( ?? ) 24. Re: DB_DELETE Function in 1.4 (Alvin Austin) 25. Re: How to exit from console? (John Novack) 26. Re: realtime sipusers and rtcachefriends... bigheadache!! (kjcsb) 27. cmd Backgound problem with option m (Franz Wu) -- Message: 11 Date: Tue, 23 Jan 2007 15:10:28 -0600 From: Mike Hammett [EMAIL PROTECTED] Subject: [asterisk-users] Snom 320 echo To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070123/ef7ed599/attachment-0001.htm -- Message: 12 Date: Tue, 23 Jan 2007 14:21:52 -0700 From: Colin Anderson [EMAIL PROTECTED] Subject: RE: [asterisk-users] Snom 320 echo To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Later firmware versions have an echo-cancelling component in it, upgrade to latest version and also turn down the gains on the mic, the default setting is way too high. A setting of 3 or 4 max is all that is nessisary. hth -Original Message- From: Mike Hammett [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 23, 2007 2:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Snom 320 echo Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070123/22d41817/attachment-0001.htm -- Message: 13 Date: Tue, 23 Jan 2007 22:30:44 +0100 (MET) From: Christian Stredicke [EMAIL PROTECTED] Subject: AW: [asterisk-users] Snom 320 echo To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
[asterisk-users] Snom 320 echo
Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Network\Snom phone oddity
I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command ping -f -c 1 192.168.2.10. Outgoing calls are junk, incoming calls are fine. (relatively speaking) The config from one phone to the next is the same except for account and voicemail settings. sip.conf is the same except for account. okay, the phone is bad, so I order a new one. This phone, however, is reporting 4% - 30% packet loss so every call is horrible just due to the lost packets (I'd assume). I install a new cable into a different port on the switch (same port as a working phone, with the working phone going into the same port as the old cable). Same results. Take this phone elsewhere. Packet loss continues. I even try different power supplies and handsets to find SOME sort of fault other than the obvious. I take the old phone back to my office and it works flawlessly, though my client uses the phone constantly all day whereas we only did approximately a half hour of testing. I take the new phone back to my office and it now has 0% packet loss. So, do I have two broken phones or is there something else wrong? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor or log peer performance
In a couple different locations I have some clients that are having intermittent problems. All of my other customers aren't complaining of issues. Whenever I conduct a test, everything is fine. No call quality issues to speak of. What can I do to log\monitor these clients so I can troubleshoot this issue? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASTTAPI
Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom high SIP ping time
I don't know everything that's going on as someone else has been working on the project, but it hasn't really been going anywhere, so I had some questions. We've got some Snom 320s with Asterisk 1.2.9.1 (I believe). All was well (with a previous release), but the phones started to get real choppy. We are also running a softphone at this location and it was fine. The SIP qualify was returning ping times anywhere from 20 to 70 ms over a sparsely used LAN. Command prompt (ICMP) pings were under 1 ms. No amount of different Asterisk versions or phone firmware revisions seems to solve this. All was well, then (as far as we know) without changes, it crapped out. Any ideas? Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Snom high SIP ping time
Well, it wasn't so much of a command line ping on the SIP port, but the times reported under Status when qualify is set to yes. It should be far less time than that as the time from that PBX up to my server out on the public Internet is only 10 ms away. I have servers in other parts of the country that are only 55 ms away. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com If you ping on the SIP port the message has to go through the application layer - which takes some time considering it is an embedded system with a small CPU. That part should be ok. It the phone becomes choppy, that problem is probably related to the RTP side. Maybe you have different packet sizes for incoming and outgoing traffic. You can get an Ethereal trace from the web interface of the phone which should show you the RTP jitter (PCAP trace). Or use a hub if you don't trust that trace. 6.1 is the latest version if you want to try the latest image (http://www.snom.com/wiki/index.php/Beta_Firmware). Hope that helps, CS ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX multiport ATA
I'm looking for an ATA\Voice Gateway that runs IAX and has several ports (8 would be nice). I am looking to avoid devices that use the same firmware as the ATCOM devices as I found them to be buggy (and a PITA to find the proper update). --Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Diverse servers
I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system such that in event of failure, childAsterisk boxes, phones, ATAs, etc. can register to either box. I can handle the child's configuration, but how do I have it setup on the Asterisk boxes? I'm not exactly sure I explained this right, but hopefully someone can get what I'm talking about and ask further questions of me. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Diverse servers
I guess my question is... I have servers A and B. They are not at the same location, but 30 - 60 ms apart. They do not have the same capacities. Those servers connect to various upstreams with the same login credentials. An IAXy can register to both servers, but only registers to one at a time. Let's say its currently registered to B. If a call comes in on A, how do we direct it to the IAXy via server B without removing the possibility that the IAXy registers to server A. Now there's servers A through H. In addition to IAXys, there's client Asterisk systems, SIP phones, etc. Next step? Geographically diverse servers, and I'm afraid of a call coming in to a server that don't know what to do with it when another server knows exactly what to do with it. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, May 17, 2006 9:09 AM Subject: Asterisk-Users Digest, Vol 22, Issue 95 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Delay when ringing internal extensions on incoming zap call (Derek Lee-Wo) 2. RE: DELL PowerEdge 2850 and TE4110P and TE110P (Asterisk) 3. Re: tdm2400p: fax detection not working (Kevin P. Fleming) 4. Re: DELL PowerEdge 2850 and TE4110P and TE110P (Julian Lyndon-Smith) 5. Re: CallerID retain on internal transfer (Kevin P. Fleming) 6. Re: Plan to free myself from AAH (John Novack) 7. Reading queue_logs ([EMAIL PROTECTED]) 8. Re: WiFi VoIP Handsets.. (Andrew Latham) 9. Re: NO ringing tone while dialing (Philipp von Klitzing) 10. (no subject) (Jordan Novak) 11. IAX crackilng (Jordan Novak) 12. Re: Using REGEX function (Kevin P. Fleming) 13. Re: IAX crackilng (Rich Adamson) 14. Re: SIP Min-Expires (Kevin P. Fleming) 15. Diverse servers (Mike Hammett) 16. WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic! (Edu) 17. Re: Sangoma A200D problem (Andre Courchesne - Consultant) 18. Re: WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic! (Edu) 19. RE: Diverse servers (Brian C. Fertig) 20. RE: WiFi VoIP Handsets.. (Cory Andrews) 21. Re: WiFi VoIP Handsets.. (Colin MacMillan) -- Message: 19 Date: Wed, 17 May 2006 09:56:06 -0400 From: Brian C. Fertig [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Diverse servers To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii For your configuration to be like this RRDNS and Realtime. I believe someone made a patch for realtime to work correctly with RRDNS you would have to check the wiki or mantis to find it. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Wednesday, May 17, 2006 9:51 AM To: Asterisk-users@lists.digium.com Subject: [Asterisk-Users] Diverse servers I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system such that in event of failure, child Asterisk boxes, phones, ATAs, etc. can register to either box. I can handle the child's configuration, but how do I have it setup on the Asterisk boxes? I'm not exactly sure I explained this right, but hopefully someone can get what I'm talking about and ask further questions of me. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. -- next part -- An HTML attachment
[Asterisk-Users] Dial plan question - exclamtion mark
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns says: ! wildcard, matches zero or more characters immediately (only Asterisk 1.2 and later, see note) Note: The exclamation mark wildcard, which is available only in Asterisk 1.2 and later, behaves specially it will match as soon as can without waiting for the dialing to complete, but it will not match until it is unambiguous, and the number being dialed cannot match any other extension in the context. It was designed for use as follows, so that as soon as the digits dialed don't match '001800...' the outgoing telephone line will be picked up and overlap dialing will be used (with full audio feedback from 'earlyb3' etc.) Context "outgoing": Extension Description _001800NXX Free US calls made by VoIP _X! Outgoing calls via normal telco, with overlap dial. = So then can I have _!800NXX to match someone dialing 18005551212 and 8005551212? If not, what could I do in this situation? Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trunking Questions
Is there a non hardware limit to the limit of concurrent connections that can go over a trunk? So IAX trunking is preferred, can * do any other trunking? Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] trunking questions
Is there a non hardware limit to the limit of concurrent connections that can go over a trunk? So IAX trunking is preferred, can * do any other trunking? Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDMoE
Well, I don't know what it is at the moment, I just know its a wireless T-1 that I'd migrate over to a different infrastructure. Actually, TDMoE can route and can go longer distances when you run it over Mikrotik and use their EoIP. Well, given that the fact that it runs over Ethernet instead of IP is its only issue. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 09, 2006 3:11 AM Subject: Asterisk-Users Digest, Vol 19, Issue 59 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Asterisk-Users Digest, Vol 19, Issue 58 (Mike Hammett) 2. RE: Re: Asterisk-Users Digest, Vol 19, Issue 58 (Alexander Lopez) 3. RE: Two Lines, Two Businesses (Les Bell) 4. Re: Welltech USA? and Wellgate Products? (Dinesh Nair) 5. Re: ([EMAIL PROTECTED]) 6. Re: Two Lines, Two Businesses ([EMAIL PROTECTED]) 7. NSLU2 Asterisk (sukrit) 8. What ATA should I buy? (Tomislav Par?ina) 9. Queue - joinempty (Tomislav Par?ina) 10. RE: Two Lines, Two Businesses (Alexander Lopez) 11. Fax transmission interrupt on ISDN network (Olivier Krief) 12. Voicemail Problem (Sam Lee) 13. Re: ztdummy on gentoo 2005.1 (Tzafrir Cohen) 14. Voicemailmain() refusing connection problem (Sam Lee) 15. Tormenta 2 and channel bank (Viktor Tatianin) 16. TDM400p (Hans Witvliet) 17. Re: Web based SIP client (Klaus Darilion) 18. How can I send DTMF from the console? (Anthony Azzopardi) 19. RE: cisco 7940 firmware upgrade (kevin ling) 20. Re: Bandwidth: to seperate or not to seperate (Derek Conniffe) 21. RE: festival-script.pl... howto change language? (kevin ling) -- Message: 1 Date: Thu, 9 Feb 2006 00:20:14 -0600 From: Mike Hammett [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 19, Issue 58 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=original Reason I ask is I may have a non-voice T-1 replacement project going on and I'm investigating my various options. Costs may be about the same for turn-key and DIY. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 09, 2006 12:07 AM Subject: Asterisk-Users Digest, Vol 19, Issue 58 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Welltech USA? and Wellgate Products? (kevin ling) 2. RE: Connecting to live calls (Wai Wu) 3. RE: Web based SIP client (kevin ling) 4. Re: 911 and ISDN PRI (Darren Nickerson) 5. Asterisk returning 403 Forbidden response ([EMAIL PROTECTED]) 6. RE: Connecting to live calls (Alexander Lopez) 7. TDMoE (Mike Hammett) 8. SIP-H323 Help and Multiple Listening Port (Kenige Ho) 9. RE: TDMoE (Alexander Lopez) 10. Re: Mitel 5220 IP phones (tracinet) 11. Polycom dialplan restriction (Carlos Chavez) 12. SER + Asterisk (Nick Hoffman) 13. OOH323 Configuration (Abdul Lateef) 14. Re: Bandwidth: to seperate or not to seperate (Rich Adamson) 15. RE: PRI indications. (Mark Edwards) -- Message: 9 Date: Wed, 8 Feb 2006 23:59:18 -0500 From: Alexander Lopez [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] TDMoE To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii TDM is not limited to voice. But there are better ways of moving data across an ethernet segment. Look at the various treads recently about TDMoE. Make sure you are using a separate card for anytype of non-testing load. Use a 2.6 based kernel, Better networking. Pick a religion and follow it, you with need a bit a divine intervention. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Wednesday, February 08, 2006 11:43 PM To: asterisk-users@lists.digium.com Subject
[Asterisk-Users] TDMoE
Can TDMoE be used for non-voice applications? Can another box be setup with TDMoE on the other side to dump it back out via T-1? How does this compare with an off-the-shelf TDM over Ethernet or IP device? Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 19, Issue 58
Reason I ask is I may have a non-voice T-1 replacement project going on and I'm investigating my various options. Costs may be about the same for turn-key and DIY. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 09, 2006 12:07 AM Subject: Asterisk-Users Digest, Vol 19, Issue 58 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Welltech USA? and Wellgate Products? (kevin ling) 2. RE: Connecting to live calls (Wai Wu) 3. RE: Web based SIP client (kevin ling) 4. Re: 911 and ISDN PRI (Darren Nickerson) 5. Asterisk returning 403 Forbidden response ([EMAIL PROTECTED]) 6. RE: Connecting to live calls (Alexander Lopez) 7. TDMoE (Mike Hammett) 8. SIP-H323 Help and Multiple Listening Port (Kenige Ho) 9. RE: TDMoE (Alexander Lopez) 10. Re: Mitel 5220 IP phones (tracinet) 11. Polycom dialplan restriction (Carlos Chavez) 12. SER + Asterisk (Nick Hoffman) 13. OOH323 Configuration (Abdul Lateef) 14. Re: Bandwidth: to seperate or not to seperate (Rich Adamson) 15. RE: PRI indications. (Mark Edwards) -- Message: 9 Date: Wed, 8 Feb 2006 23:59:18 -0500 From: Alexander Lopez [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] TDMoE To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii TDM is not limited to voice. But there are better ways of moving data across an ethernet segment. Look at the various treads recently about TDMoE. Make sure you are using a separate card for anytype of non-testing load. Use a 2.6 based kernel, Better networking. Pick a religion and follow it, you with need a bit a divine intervention. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Wednesday, February 08, 2006 11:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TDMoE Can TDMoE be used for non-voice applications? Can another box be setup with TDMoE on the other side to dump it back out via T-1? How does this compare with an off-the-shelf TDM over Ethernet or IP device? Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060208/7b569589/attachment-0001.htm -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple call groups
As evident in the SuperDial script and others based upon groups, you can place a call into a group, which can have a limit on the number of concurrent calls. Can a call belong to multiple groups? IE: I have only a limited number of channels to upstream X. Downstream Y is only paying me for a limited number of channels. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
