Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Sure thing! Bug #18302 has been opened (https://issues.asterisk.org/view.php?id=18302). Brett Woollum br...@woollum.com - Original Message - From: Sherwood McGowan sherwood.mcgo...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 12:20:23 PM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables Sounds good mate, keep me posted, and let me know the issue number so I can check in on it :D Who knows, I might be able to offer some testing or somethin' for the digium guys or something -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan and sip
Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan and sip
What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Hi All, I'm having an issue where Asterisk continuously sends out a GAZILLION SIP NOTIFY messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. I'd like to confirm the layout of the sip_users table in my MySQL database to make sure it's what Asterisk is looking for. I cannot find any official documentation that specifies what the sip_users table should look like. Their documentation system does a great job of showing what the table should look like for the Voicemail ODBC storage, for example. But not for the Realtime sip_users table. I'm currently using the table listed here: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip Is there any official documentation on this? Brett Woollum br...@woollum.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Hi Paul, 1.6.2.13. I'll go ahead and update to 1.6.2.14 and see how that works. I did see a couple bugs in the bug tracker for this, but they were resolved a while ago (I want to say 1.6.1 timeframe...). There was also a post on this list about the problem arising from LDAP integration, but I didn't see any resolution posted. Brett Woollum br...@woollum.com - Original Message - From: Paul Belanger paul.belan...@polybeacon.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 4:58:24 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables On Fri, Nov 12, 2010 at 6:07 AM, Brett Woollum br...@woollum.com wrote: I'm having an issue where Asterisk continuously sends out a GAZILLION SIP NOTIFY messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. What version of 1.6? I _think_ this may have been a bug, that was fixed. Don't hold me to that. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Hi Brad, I did notice that bug in the bug tracker. That's different from the behavior I am seeing. I don't get multiple values in the Mailbox. I just upgraded to 1.6.2.14 and it's still there. By the way, the quantity of SIP NOTIFY's generated is significant. It appears to be way more that the number of peers I have (3) times a handful of duplicates per peer. I've been doing a Wireshark capture, and it appears as though any time there is a new message in the ODBC voicemail store for a mailbox that has been subscribed to, Asterisk continually generates as many of the messages as possible. At one point I noticed my CPU jump from 0% to ~50% just by moving one message from an mailbox that hadn't been subscribed to to a mailbox that was subscribed to by the 3 peers. It only came back to ~0-1% by moving the message back to an unsubscribed user. When I set rtcachefriends = yes in sip.conf, I get the following for each peer: ast01*CLI sip show peer 412 * Name : 412 Realtime peer: Yes, cached Secret : Set MD5Secret : Not set Remote Secret: Not set Context : sipphones Subscr.Cont. : blf_subscriptions Language : en AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : vm_...@default VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : MaxCallBR : 384 kbps Expire : 69 Insecure : no Nat : RFC3581 ACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes Forward Loop : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 10.20.1.225 Port 5064 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 412 SIP Options : (none) Codecs : 0x1004 (ulaw|g722) Codec Order : (g722:20,ulaw:20) Auto-Framing : No 100 on REG : Yes Status : Unmonitored Useragent : Yealink SIP-T28P 2.50.0.52 Reg. Contact : sip:4...@10.20.1.225:5064 Qualify Freq : 12 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs Parkinglot : This is Asterisk 1.6.2.14 using the ODBC store for voicemail and ODBC for sip_peers. Brett Woollum br...@woollum.com - Original Message - From: Bradley Watkins bradley.watk...@compuware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 5:14:49 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Friday, November 12, 2010 7:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables On Fri, Nov 12, 2010 at 6:07 AM, Brett Woollum br...@woollum.com wrote: I'm having an issue where Asterisk continuously sends out a GAZILLION SIP NOTIFY messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. What version of 1.6? I _think_ this may have been a bug, that was fixed. Don't hold me to that. I agree with Paul, this sounds like a bugs that's been fixed. What does the 'Mailbox :' line look like when you do a 'sip show peers'? My guess is that there will be multiple entries of the same mailbox, and that's why you're receiving a bunch of NOTIFY messages. - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
More information: When I have rtcachefriends = yes in sip.conf, everything seems fine. With rtcachefriends = no I see this behavior. I'd rather not cache. I'm aiming for as near real-time as possible. Any thoughts? Brett Woollum br...@woollum.com - Original Message - From: Brett Woollum br...@woollum.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 5:34:03 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables Hi Brad, I did notice that bug in the bug tracker. That's different from the behavior I am seeing. I don't get multiple values in the Mailbox. I just upgraded to 1.6.2.14 and it's still there. By the way, the quantity of SIP NOTIFY's generated is significant. It appears to be way more that the number of peers I have (3) times a handful of duplicates per peer. I've been doing a Wireshark capture, and it appears as though any time there is a new message in the ODBC voicemail store for a mailbox that has been subscribed to, Asterisk continually generates as many of the messages as possible. At one point I noticed my CPU jump from 0% to ~50% just by moving one message from an mailbox that hadn't been subscribed to to a mailbox that was subscribed to by the 3 peers. It only came back to ~0-1% by moving the message back to an unsubscribed user. When I set rtcachefriends = yes in sip.conf, I get the following for each peer: ast01*CLI sip show peer 412 * Name : 412 Realtime peer: Yes, cached Secret : Set MD5Secret : Not set Remote Secret: Not set Context : sipphones Subscr.Cont. : blf_subscriptions Language : en AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : vm_...@default VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : MaxCallBR : 384 kbps Expire : 69 Insecure : no Nat : RFC3581 ACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes Forward Loop : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 10.20.1.225 Port 5064 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 412 SIP Options : (none) Codecs : 0x1004 (ulaw|g722) Codec Order : (g722:20,ulaw:20) Auto-Framing : No 100 on REG : Yes Status : Unmonitored Useragent : Yealink SIP-T28P 2.50.0.52 Reg. Contact : sip:4...@10.20.1.225:5064 Qualify Freq : 12 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs Parkinglot : This is Asterisk 1.6.2.14 using the ODBC store for voicemail and ODBC for sip_peers. Brett Woollum br...@woollum.com - Original Message - From: Bradley Watkins bradley.watk...@compuware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 5:14:49 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Friday, November 12, 2010 7:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables On Fri, Nov 12, 2010 at 6:07 AM, Brett Woollum br...@woollum.com wrote: I'm having an issue where Asterisk continuously sends out a GAZILLION SIP NOTIFY messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. What version of 1.6? I _think_ this may have been a bug, that was fixed. Don't hold me to that. I agree with Paul, this sounds like a bugs that's been fixed. What does the 'Mailbox :' line look like when you do a 'sip show peers'? My guess is that there will be multiple entries of the same mailbox, and that's why you're receiving a bunch of NOTIFY messages. - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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
Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Hi Sherwood, Thanks for the reply. That's interesting to me. What is the point of rtcachefriends = no if it causes weird things like this to happen? As mentioned, I'd like to stay real-time and fully database driven for everything. Not only does it make life easier in terms of changing settings on the system (without reloads!), but it will make scaling the system to more Asterisk servers much easier. Is there a way for Asterisk to automatically look up the sip user or peer's information from the ODBC backend every time and work properly? It seems to be doing that with rtcachefriends = no, with the exception of the MWI subsystem. How can I retain the database driven behavior of rtcachefriends = yes, but still keep the MWI working? Also, the BLF subscriptions and subsequent NOTIFY's are working fine. A capture of the wire by the phone shows the only issue as being the NOTIFY's for MWI. Thanks! Brett Woollum br...@woollum.com - Original Message - From: Sherwood McGowan sherwood.mcgo...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 7:36:22 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables On Fri, Nov 12, 2010 at 7:52 AM, Brett Woollum br...@woollum.com wrote: More information: When I have rtcachefriends = yes in sip.conf, everything seems fine. With rtcachefriends = no I see this behavior. I'd rather not cache. I'm aiming for as near real-time as possible. Any thoughts? Brett Woollum br...@woollum.com - Original Message - From: Brett Woollum br...@woollum.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 5:34:03 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables Hi Brad, I did notice that bug in the bug tracker. That's different from the behavior I am seeing. I don't get multiple values in the Mailbox. I just upgraded to 1.6.2.14 and it's still there. By the way, the quantity of SIP NOTIFY's generated is significant. It appears to be way more that the number of peers I have (3) times a handful of duplicates per peer. I've been doing a Wireshark capture, and it appears as though any time there is a new message in the ODBC voicemail store for a mailbox that has been subscribed to, Asterisk continually generates as many of the messages as possible. At one point I noticed my CPU jump from 0% to ~50% just by moving one message from an mailbox that hadn't been subscribed to to a mailbox that was subscribed to by the 3 peers. It only came back to ~0-1% by moving the message back to an unsubscribed user. When I set rtcachefriends = yes in sip.conf, I get the following for each peer: ast01*CLI sip show peer 412 * Name : 412 Realtime peer: Yes, cached Secret : Set MD5Secret : Not set Remote Secret: Not set Context : sipphones Subscr.Cont. : blf_subscriptions Language : en AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : vm_...@default VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : MaxCallBR : 384 kbps Expire : 69 Insecure : no Nat : RFC3581 ACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes Forward Loop : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 10.20.1.225 Port 5064 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 412 SIP Options : (none) Codecs : 0x1004 (ulaw|g722) Codec Order : (g722:20,ulaw:20) Auto-Framing : No 100 on REG : Yes Status : Unmonitored Useragent : Yealink SIP-T28P 2.50.0.52 Reg. Contact : sip:4...@10.20.1.225:5064 Qualify Freq : 12 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs Parkinglot : This is Asterisk 1.6.2.14 using the ODBC store for voicemail and ODBC for sip_peers. Brett Woollum br...@woollum.com - Original Message - From: Bradley Watkins bradley.watk...@compuware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 12, 2010 5:14:49 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Friday
Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables
Yeah a production system that crashes is not fun.. I hear you there. Maybe the solution will be to design some sort of method for each asterisk server to auto prune and load as necessary. The first issue that is coming to mind is that I'm doing configuration and db changes/updates on a different server than asterisk. This would mean my web server would need to reach out to each asterisk server to tell it to update. This will be interesting. I think I will try opening a bug ticket. I think a stable database backend for Asterisk is critical for easy integration with other systems and scaling the platform. Fixing MWI is just a stepping stone. As far as I can tell the rest of the Realtime architecture I've implemented works fine. Unfortunately I'm not a C coder. I never could get used to it. I actually tried to run through the code last night to see what I could find, but I didn't understand much. I was able to find some of the MWI notify functions (in chan_sip.so and events.so for example), but nothing stood out to me. I would like to work with whomever I can to try resolving this. Hopefully we can figure it out and get MWI working with rtcachefriends = no (or maybe a little hahaha)! Brett Woollum Sent from my iPhone On Nov 12, 2010, at 11:17 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Inline response :D On Fri, Nov 12, 2010 at 12:54 PM, Brett Woollum br...@woollum.com wrote: Hi Sherwood, Thanks for the reply. Most definitely mate, since I've used realtime so much, I enjoy digging in there. However, I use the MySQL realtime architecture, so forgive me if we find there's differences between the ODBC architecture's behavior and what I believe it should be :) That's interesting to me. What is the point of rtcachefriends = no if it causes weird things like this to happen? The long and short of it is, IIRC, MWI did not work with realtime until they added the realtime cache functionality. So, when you turn it off, it goes back to the good old days of 2005, when Realtime was still experimental. Well, not really, you just have weirdness with notification packets, in '05 the realtime engine would just randomly crash asterisk...that made my life hell, seeing as how I had built a certain ITSP's system with the experimental realtime because they insisted :P As mentioned, I'd like to stay real-time and fully database driven for everything. Not only does it make life easier in terms of changing settings on the system (without reloads!), but it will make scaling the system to more Asterisk servers much easier. Trust me, I'm pickin' up what you're putting down mate. Is there a way for Asterisk to automatically look up the sip user or peer's information from the ODBC backend every time and work properly? It seems to be doing that with rtcachefriends = no, with the exception of the MWI subsystem. How can I retain the database driven behavior of rtcachefriends = yes, but still keep the MWI working? Well, you can do one of two things, I reckon: 1) Submit a bug report about the excess of notify messages and whatnot and work with whoever picks up the ticket as much as possible. I did that with Murf concerning improving AEL AND rectifying the macro iteration depth issue (I was the poor bastard that discovered it, in the middle of a 3 month long development of a LARGE wholesale/reseller/ITSP project)... OR 2) Learn C (if you don't already know it), and take a crack at the code yourself. There's a couple others, but those two are the sure fire methods ;-) Also, the BLF subscriptions and subsequent NOTIFY's are working fine. A capture of the wire by the phone shows the only issue as being the NOTIFY's for MWI. Right on, I kinda figured as soon as you mentioned rtcachefriends. We're basically stuck at The RealTime engine has had this issue since 2005/2006, and there's been no massive complaints about this... I don't like it, I'd personally like MWI to work without caching, or maybe only caching a LITTLE bit of data. One other quick note though, there's a good reason to not be COMPLETELY realtime with your SIP or IAX clients' configurations...That would mean that Asterisk would have to query the database for EVERY realtime client configuration every time it needs to do a MWI check, which is probably why it would crash back in the days of me getting 1-2 hours sleep in 24-48 hours constantly... Sorry I can't do anything more than explain what I know, but I've never delved into it because even in my clustering/HA setups, I've just dealt with doing the prune and loads. Keep in mind, the prune and load method only performs the action on THE SPECIFIC CLIENT you request it for, it's not a sip reload :D Slainte Mate! Sherwood Thanks! Brett Woollum br...@woollum.com - Original Message - From: Sherwood McGowan sherwood.mcgo...@gmail.com To: Asterisk Users Mailing List - Non-Commercial
Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
That was it! I had a value (412 and 413) set for each phone. This overwrote the
caller ID that I was setting in the dialplan. Removing the contents of the
fromuser field cleared this issue.
Thanks Olle!
Brett Woollum
br...@woollum.com
- Original Message -
From: Olle E. Johansson o...@edvina.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 11:30:27 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
10 nov 2010 kl. 02.38 skrev Brett Woollum:
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412, 3)
exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1)
exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to 2.
SIP/412 is dialed next. It receives the call, but with 412 as the Caller ID
number - even though the real source of the call was extension 413. The name
I set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413
into sip.conf directly, this code works (ie: the CallerID(num) I set makes it
out to the destination phone properly).
Have you set the fromuser= field in the realtime database?
/O
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Hi Carlos.
Yes I did have fromuser set, which was the problem. I removed this for each
extension and that solved the issue.
Thanks!
Brett Woollum
br...@woollum.com
- Original Message -
From: Carlos Chavez cur...@telecomabmex.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 10, 2010 8:47:38 AM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, 2010-11-09 at 17:38 -0800, Brett Woollum wrote: Good idea Paul.
My debug output: [Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP
CoS mark 5 [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2) in new stack [Nov 9
17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack [Nov 9 17:33:39]
VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5 [Nov 9 17:33:39]
VERBOSE[4175] app_dial.c: -- Called 412 [Nov 9 17:33:40] VERBOSE[4175]
app_dial.c: -- SIP/412-0006 is ringing [Nov 9 17:33:44] VERBOSE[4175]
pbx.c: == Spawn extension (sipphones, 412, 3) exited non-zero on
'SIP/413-0005' [Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing
[...@sipphones:1] Hangup(SIP/413-0005, ) in new stack [Nov 9
17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1) exited
non-zero on 'SIP/413-0005' As you can see on line 3, CallerID(num) was
successfully set to 2. SIP/412 is dialed next. It receives the call,
but with 412 as the Caller ID number - even though the real source of the
call was extension 413. The name I set in CallerID(name) works fine. My
Extensions.conf for that context: [sipphones] exten =
412,1,Set(CALLERID(num)=2) exten =
412,1,Set(CALLERID(all)=TEST2) exten = 412,n,NoOp(CallerID(num) is:
${CALLERID(num)}) exten = 412,n,Dial(SIP/412) exten =
412,n,NoOp(${CALLERID(num)}) If I disable sippusers and sippeers in
extconfig.conf and put 412 and 413 into sip.conf directly, this code works
(ie: the CallerID(num) I set makes it out to the destination phone properly).
Are you using the fromuser field in the realtime table? I had this problem
once when from user was set and user kept receiving that as the callerid. --
Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director
de Tecnología +52-55-91169161 ext 2001
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Nobody has any idea why the Caller ID is being overwritten when using Asterisk
Realtime for the SIP users?
Brett Woollum
br...@woollum.com
- Original Message -
From: Brett Woollum br...@woollum.com
To: asterisk-users@lists.digium.com
Sent: Sunday, November 7, 2010 3:08:50 PM GMT -08:00 US/Canada Pacific
Subject: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num)
Problem
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem setting the Caller ID number
whenever I am using the Realtime database for the SIP users/peers. If I use a
static sip.conf configuration instead of the database, everything works fine.
Unfortunately a static sip.conf file won't work in my application.
In this example:
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)}) ;;;PS: This shows the
correct number of 2 on the CLI console...
exten = 412,n,Dial(SIP/412)
Whenever another phone calls extension 412, the call is forwarded to SIP/412
and should have TEST as the CallerID name and 2 as the CallerID number.
But, whenever I am using the realtime backend, the caller ID number always
displays on the destination phone as that phone's username. Meaning, if phone
SIP/412 receives the call from the example above, the caller ID name displayed
is TEST but the caller ID number is always 412.
What could be causing this?
Brett Woollum
br...@woollum.com
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Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set(SIP/413-0005, CALLERID(num)=2) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:2]
NoOp(SIP/413-0005, CallerID(num) is: 2 ) in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:3]
Dial(SIP/413-0005, SIP/412) in new stack
[Nov 9 17:33:39] VERBOSE[4175] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] app_dial.c: -- Called 412
[Nov 9 17:33:40] VERBOSE[4175] app_dial.c: -- SIP/412-0006 is ringing
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, 412, 3)
exited non-zero on 'SIP/413-0005'
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Hangup(SIP/413-0005, ) in new stack
[Nov 9 17:33:44] VERBOSE[4175] pbx.c: == Spawn extension (sipphones, h, 1)
exited non-zero on 'SIP/413-0005'
As you can see on line 3, CallerID(num) was successfully set to 2.
SIP/412 is dialed next. It receives the call, but with 412 as the Caller ID
number - even though the real source of the call was extension 413. The name I
set in CallerID(name) works fine.
My Extensions.conf for that context:
[sipphones]
exten = 412,1,Set(CALLERID(num)=2)
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)})
exten = 412,n,Dial(SIP/412)
exten = 412,n,NoOp(${CALLERID(num)})
If I disable sippusers and sippeers in extconfig.conf and put 412 and 413 into
sip.conf directly, this code works (ie: the CallerID(num) I set makes it out to
the destination phone properly).
Brett Woollum
br...@woollum.com
- Original Message -
From: Paul Belanger paul.belan...@polybeacon.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 9, 2010 5:18:36 PM GMT -08:00 US/Canada Pacific
Subject: Re: [asterisk-users] Asterisk 1.6 (realtime) - Overwritten
CallerID(num) Problem
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum br...@woollum.com wrote:
Nobody has any idea why the Caller ID is being overwritten when using
Asterisk Realtime for the SIP users?
No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger
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[asterisk-users] Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem setting the Caller ID number
whenever I am using the Realtime database for the SIP users/peers. If I use a
static sip.conf configuration instead of the database, everything works fine.
Unfortunately a static sip.conf file won't work in my application.
In this example:
exten = 412,1,Set(CALLERID(all)=TEST2)
exten = 412,n,NoOp(CallerID(num) is: ${CALLERID(num)}) ;;;PS: This shows the
correct number of 2 on the CLI console...
exten = 412,n,Dial(SIP/412)
Whenever another phone calls extension 412, the call is forwarded to SIP/412
and should have TEST as the CallerID name and 2 as the CallerID number.
But, whenever I am using the realtime backend, the caller ID number always
displays on the destination phone as that phone's username. Meaning, if phone
SIP/412 receives the call from the example above, the caller ID name displayed
is TEST but the caller ID number is always 412.
What could be causing this?
Brett Woollum
br...@woollum.com
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