[Asterisk-Users] Problem redirecting to voicemail through a SIP proxy (Looks like a bug)
I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail. GATEWAY---PROXY---FEATURE | | UA For simplicity, hostnames and IPs replaced with the above names. USERNAME, DSTNUM and SRCNUM also used to replace the UA's username, the source number of the call, and the destination number of the call. The basic SIP dialog goes: Gateway invites proxy proxy invites UA UA replies 180 Ringing. (Transaction times out and drops to failure route) PROXY invites FEATURE server (INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0.) PROXY cancels UA FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY UA tells PROXY '487 Request Terminated.' FEATURE replies 200 ok to PROXY PROXY replies 200 ok to GATEWAY ... For some reason, the asterisk gateway doesn't seem to be ACKing the 200 ok. I don't see any new invite or reinvite going to the gateway which I think may be what is confusing it. (I know this is the asterisk users list, but here is a Ser.cfg excerpt for anyone with experience with both. ) ## # User did not answer phone, or could not connect, or is on the phone and does not use call waiting. ## failure_route[3] { if(isflagset(10)) { if(t_check_status(486)) { if (!subst_user('/^/*voicemail-busy-/')){ log(1,Err in subst_user\n); } xlog(L_ERR, Relaying to voicemail Busy\n); } else { if (!subst_user('/^/*voicemail-noanswer-/')){ log(1,Err in subst_user\n); } xlog(L_ERR, Relaying to voicemail No answer\n); } rewritehostport(FEATURE:5060); append_branch(); t_relay(); } } From the gateway's point of view, the invite looks like 1. U GATEWAY:5060 - PROXY:5060 2. INVITE sip:[EMAIL PROTECTED] SIP/2.0. 3. Via: SIP/2.0/UDP GATEWAY:5060;branch=z9hG4bK6e117757. 4. From: SRCNUM sip:[EMAIL PROTECTED];tag=as7f56ca42. 5. To: sip:[EMAIL PROTECTED]. 6. Contact: sip:[EMAIL PROTECTED]. 7. Call-ID: [EMAIL PROTECTED] 8. CSeq: 102 INVITE. 9. User-Agent: Asterisk PBX. 10. Date: Tue, 20 Sep 2005 21:13:57 GMT. 11. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. 12. Content-Type: application/sdp. 13. Content-Length: 270. And the 200 ok looks like 1. U PROXY:5060 - GATEWAY:5060 2. SIP/2.0 200 OK. 3. Via: SIP/2.0/UDP GATEWAY:5060;branch=z9hG4bK6e117757. 4. Record-Route: sip:PROXY;ftag=as7f56ca42;lr. 5. From: SRCNUM sip:[EMAIL PROTECTED];tag=as7f56ca42. 6. To: sip:[EMAIL PROTECTED];tag=as7f1b7210. 7. Call-ID: [EMAIL PROTECTED] 8. CSeq: 102 INVITE. 9. User-Agent: Asterisk PBX. 10. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. 11. Contact: sip:[EMAIL PROTECTED]. 12. Content-Type: application/sdp. 13. Content-Length: 270. See links these links for full sip dialog. Ngrep from PROXY's point of viewhttp://pastebin.ca/23469 Ngrep from GATEWAY's point of view http://pastebin.ca/23470 Ngrep from FEATURE's point of view http://pastebin.ca/23471 ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0.9 long term stability --threadhijack, why not reboot?
Just reboot is a bad attitude. If there is a memory leek, the fact that a reboot will free the leaked memory is not a good reason to not fix the memory leek. That kind of attitude is why windows does need regular reboots. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, September 14, 2005 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.0.9 long term stability --threadhijack, why not reboot? Yes.. because if a reboot is needed it isn't setup correctly. Reboots are a windows things. A correctly setup Linux server should never need rebooted. On 9/14/05, Colin Anderson [EMAIL PROTECTED] wrote: Disclaimer: Not a troll I'm curious as to this obsession with uptime is. All of the posts of this type are along the lines of After X days, Y thing does not work but if I reload or reboot, it's OK - so why not cron a reboot? Is it considered bad form or something like that? I reboot every night whether it is needed or not, not afraid to admit it, and everything works fine for me. We also do the Sunday reboot of all of our Windows servers as well as restarting all of the critical services such as IIS , SQL, Exchange etc nightly. It helps, a lot (Exchange is a notorious memory leaker) Of course, if your install processes calls 24/7 that's a different story. However, I expect that the majority of Asterisk installs are for a 9-to-5 type of operation. We run two shifts here, and we stop processing calls at 10 PM, and start again at about 6 AM - a large window of opportunity to reboot. Why not take advantage of it? I've also heard it said, something along the lines of: If you have to reboot, your server isn't set up correctly to which I say piffle. Even NASA has rebooted the Mars probes after they land and I understand that they run VXWorks, incidentally, the same RTOS that my Mitel 3300 uses, and *even Mitel* recommends periodic reboots, which we duly cron every night, 2 AM. 24/7/365 installs aside, is there a reason why reboots seem to be frowned upon? Again, not trolling, just curious. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys PAP2-NA failures...
I've had some of these fail with the red light. I think customers may be plugging them into the wall with a pstn line still connected. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Craig Guy Sent: Tuesday, July 19, 2005 11:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys PAP2-NA failures... I have found my PAP2-NA's to be picky about their DHCP server. My PAP2-NA appears to lockup if it is set for DHCP and the server is my Netgear RP614 Websafe router. The fix for this is to unplug the ethernet from the unit, plug in an analog handset, power it on, using the handset perform a factory reset and then set it for static IP settings. Craig - Original Message - From: Kris Stark [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 20, 2005 6:24 AM Subject: [Asterisk-Users] Linksys PAP2-NA failures... Has anybody else experienced problems with the Linksys PAP2-NA's? I've now had two of them fail unexpectedly, with no apparent rhyme or reason, having gone into a RED power LED, with a solid blue ethernet LED. No response from the device either on the network or from the phone To make matters even crazier, the one that now failed was the one I received as a replacement for the previous dead one - and no, they were never installed in the same location either Grrr Kris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Urgent Help neededt!! Asterisk 1.0.7 CPU at 99%
This is usually a mpeg123 problem. try removing the moh module and see if it goes away. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sebastian Silva Sent: Monday, May 30, 2005 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Urgent Help neededt!! Asterisk 1.0.7 CPU at 99% Hi, Are you sure the process consuming your CPU is Asterisk? Did you tried with different codecs? Andres Maduro wrote: Hi, I am using Asterisk 1.0.7 that comes with [EMAIL PROTECTED] 1.0 ISO. I have installed chan_unicall.c and MFCR2 support with latest Steve Underwood code unicall-0.0.2pre2 (yes this is the latest version, not 0.0.2pre19!!) Asterisk process is keeping the cpu at 99% most of the time. Sebas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Icecast
There is info on the wiki about moh working with streaming audio. I was able to get that to work with icecast. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Shidan Sent: Wednesday, May 11, 2005 11:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Icecast Hi, does anyone know of * being used with icecast in any way. Does * support vorbis at all? can anyone who is working on this give me a shout. Shidan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Writing To Multiple MySql Tables
Use cdr-mysql and odbc interfaced to a mysql database -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rafal Kaniewski Sent: Tuesday, May 10, 2005 5:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] RE: Writing To Multiple MySql Tables Ive got realtime and mysql.cmd to read from databases but apart from cdr how else can * write? I need to write to 2x tables and cant do this with cdr? Any advice appreciated thanks. --- Rafal Kaniewski -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.7 - Release Date: 09/05/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Difference between a TE410P and TE405P?
Just the voltage -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ed Greenberg Sent: Friday, February 18, 2005 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Difference between a TE410P and TE405P? Can anybody tell me the difference between a TE410P and a TE405P? Is it JUST the 5v vs 3.3v pcis slot spec, or is there some thing else? /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Notify PAP2-NA?
I am using mysql sipfriends and can't seem to get the MWI to work. From what I've read it seems this is not supported with that dynamic system, and probably never will be. I was thinking of just setting a cron job or something to check every minute for voicemail and set our sip NOTIFY messages as needed. Also, the PAP2-NA has the ability to reboot via a sip notify and I would like to be able to do that. I have seen something to do this on some soft phones, but have not been able to get it to work on the PAP2. Anyone have any experience getting MWI to work with dynamic sipfriends or sending custom sip notify messages to PAP2's? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users