[asterisk-users] Disappearing Voicemail
I over the last 2 weeks have had voicemail start to disappear from the system. The symptoms work something like this: The user logs into Comedian Mail, and checks how many messages they have. They then opt to call back in later to check the messages, but when they do so the messages are gone, with either no messages in the box, or messages in the box having been added since the voicemail was last checked. This is occurring with Asterisk 1.2.10, and I will be updating the system to 1.2.12.1 after business hours. Has anyone seen this issue before, and if so what is the resolution? Andrew ___ --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 show peers
Response below -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Rousse Sent: Thursday, September 14, 2006 10:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip show peers Hello guys, Is there anyone who could explain me some stuff about sip show peers ? 108/10810.1.1.40 5060 OK (1 ms) 107/10710.1.1.246 D 51074OK (101 ms) The port seems different here, and the main difference is that the extension 108, is a server with a fixed IP 107, is a client with a softphone (X-Lite) and a dynamic IP. Why the diffrence in the port ? And why the difference in the reponse time ? We are on the same physical network, a ping is giving me a response of 1ms for each. Is it because the softphone is with a dynamic IP and Asterisk is treating this differently ? Thanks, SNIP A SIP ping and an ICMP ping are two different entities. The SIP ping operates at a higher level in the OSI stack than a simple ICMP ping. This means that whatever is receiving the ping has to do more work to decode it, and respond. I wouldn't worry about the latency difference as the SIP Ping is prioritized a bit more by a computer which is multitasking than by a hard phone which is not. As for the port, they simply chose to negotiate on a higher port. You might check your X-Lite settings, but I don't think this will break anything! Andrew ___ --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] Disappearing Voicemail
There is no voicemail in the old folder, I've manually inspected the folders where this is occurring via 'ls'. The mail is in fact gone. Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Thursday, September 14, 2006 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Disappearing Voicemail Andrew Kirch wrote: I over the last 2 weeks have had voicemail start to disappear from the system. The symptoms work something like this: The user logs into Comedian Mail, and checks how many messages they have. They then opt to call back in later to check the messages, but when they do so the I believe that mail that has been listened to, automatically get moved into the Old folder. The users need to select the Old mail box to listen to it again. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] RE: [asterisk-dev] Phone status
Umm Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQLtable. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: Status Privilege: Call Channel: SIP/310-08697fb8 CallerID: 310 CallerIDName: unknown Account: State: Up Link: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: Status Privilege: Call Channel: SIP/311-0868fd98 CallerID: 311 CallerIDName: Snom Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 13 Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: Status Privilege: Call Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: unknown Account: State: Up Link: IAX2/MR-1 Uniqueid: 1156442974.76 Event: Status Privilege: Call Channel: IAX2/MR-1 CallerID: 35436121 CallerIDName: unknown Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --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 version: 1.2.9.1 or older?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrea Spadaccini Sent: Wednesday, July 12, 2006 9:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk version: 1.2.9.1 or older? Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? I would suggest 1.2.9.1 as it is a security update release (ie something that can compromise your PBX is fixed); however the correct answer to this is that you need to test and determine what suits your needs. Andrew ___ --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] Phones cutting out.....again - PLEASE HELP!!!
-Original Message- Thanks for reading, Wes ___ Please reply with the output of the following: lspci -vv lspci -vv | grep IRQ lspci cat /proc/interrupts Thank you. Andrew ___ --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] intel vs amd motherboards
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Don Sent: Wednesday, July 05, 2006 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] intel vs amd motherboards If you want to handle, lets say 1000 calls or more at the same time, you should of course use a better processor. In my opinion, it doesn't matter whether you use Intel or AMD, because you said it will be a small Asterisk. In the world of asterisk...Intel or AMD really doesn't make a difference However AMD can do more for less money... I think you should concentrate more on a descent mainboard for whichever powerplant you chose to shove in it... Due to the recent nightmares I've had with the Asus K8N and the Dell PowerEdge 830, recommendations here would be greatly appreciated. What board, what interfaces (digium/sangoma) and results as well as caveats. Andrew ___ --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: Out of Office Auto Reply:
I'd second this motion, this is very very annoying. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Friday, June 23, 2006 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: Out of Office Auto Reply: Should be part of the FAQ for the list, as well as the setting for Exchange 5.5 which a *lot* of orgs still run (we do too) I wonder if the list SW can be modded to automatically plonk any mail with the subject string: Out of Office -Original Message- From: Steven [mailto:[EMAIL PROTECTED] Sent: Friday, June 23, 2006 8:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Out of Office Auto Reply: Exchange changes http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp -- -- Steven http://www.glimasoutheast.org Koopmann, Jan-Peter [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote: We use MS Exchange too and, as far as I am aware, it is cognizant of mailing list headers and doesn't send OOO notices to mailing list postings. The only mailing list from which I receive my own OOO notices is one that doesn't have the proper mailing list headers set. No. Exchange does not honour Precedence headers. It has some funky way of determining what is a mailing list and what is not. It does not work very well and it has (or had) to be enabled via a registry key. If you don't do this, even Exchange 2003 will reply to some mailing lists. But it should not send this to every mail but only once day... Kind regards, JP ___ --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 ___ --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] shutting down a mysql server renders cdr_mysql dead and asterisk nolonger makes or receives calls
At approximately 3:15pm I shut down the office MySQL server to change out some hardware. Shortly after I received a call from one of two customers whose asterisk servers output CDR data to that server. They could not place or receive calls. Shortly after that I received a call from the other customer. I'm below providing output from the message log (At debug level). I don't see much of use and would greatly appreciate any help that could be given. Andrew Jun 9 15:16:05 ERROR[29101] cdr_addon_mysql.c: cdr_mysql: Unknown connection error: (2013) Lost connection to MySQL server during query Jun 9 15:29:49 ERROR[29382] cdr_addon_mysql.c: cdr_mysql: Unknown connection error: (2013) Lost connection to MySQL server during query Jun 9 15:31:51 WARNING[4918] channel.c: Avoided initial deadlock for '0x2aaab35009c0', 10 retries! Jun 9 15:31:51 WARNING[4918] channel.c: Avoided initial deadlock for '0x2aaab35009c0', 10 retries! Jun 9 15:32:58 ERROR[29382] cdr_addon_mysql.c: cdr_mysql: cannot connect to database server 208.64.32.55. Jun 9 15:36:08 ERROR[29647] cdr_addon_mysql.c: cdr_mysql: cannot connect to database server 208.64.32.55. Jun 9 15:36:08 NOTICE[4928] pbx.c: Cannot find extension context 'did-incomig' Jun 9 15:39:41 ERROR[29796] cdr_addon_mysql.c: cdr_mysql: cannot connect to database server 208.64.32.55. Jun 9 16:19:21 NOTICE[22239] cdr.c: CDR simple logging enabled. Jun 9 16:19:22 WARNING[22239] cdr_addon_mysql.c: MySQL database sock file not specified. Using default Jun 9 16:19:23 WARNING[22239] pbx_config.c: No closing parenthesis found? 'MeetMe(50667|Msipr}' Jun 9 16:19:23 WARNING[22239] pbx_config.c: No closing parenthesis found? 'MeetMe(31391|Msipr}' Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '50667', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 103 Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '82612', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 104 Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '61908', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 105 Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '24104', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 106 Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '83416', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 107 Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '90780', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 108 Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '77252', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 110 Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '77604', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 111 Jun 9 16:19:23 WARNING[22239] pbx.c: Unable to register extension '25647', priority 1 in 'did-incoming', already in use Jun 9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 112 ___ --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] shutting down a mysql server renders cdr_mysqldead and asterisk nolonger makes or receives calls
Not res_config_mysql cdr_addon_mysql. All it does is log call detail records. According to bug http://bugs.digium.com/view.php?id=4749 cdr_addon_mysql should not behave in this way. Therefore 1. there is no realtime DB besides ASTDB storage of SIP phones, and 2. CDR is not a life-or-death situation for asterisk, if it stops asterisk should continue and replicate later (the year-old bug cited above states it already does this). Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of William Piper Sent: Friday, June 09, 2006 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] shutting down a mysql server renders cdr_mysqldead and asterisk nolonger makes or receives calls Well, it looks like you have realtime DB. The errors appear to be because it couldn't connect to the DB to resolve the extensions that the people were entering. bp On 6/9/06, Andrew Kirch [EMAIL PROTECTED] wrote: At approximately 3:15pm I shut down the office MySQL server to change out some hardware. Shortly after I received a call from one of two customers whose asterisk servers output CDR data to that server. They could not place or receive calls. Shortly after that I received a call from the other customer. I'm below providing output from the message log (At debug level). I don't see much of use and would greatly appreciate any help that could be given. Andrew ___ --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] Speex fans?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dr. Michael J. Chudobiak Sent: Friday, May 12, 2006 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Speex fans? Hi all, I've been testing various codecs to eliminate choppiness that I sometimes get on my Asterisk IAX2 DSL provider (Exgn) connections, and Speex seems to work the best, so far - but Speex seems oddly unpopular. Can anyone share their experiences with Speex (good and bad)? Is anyone using it in a production environment? I like the variable bit rate and packet loss concealment features... - Mike I believe the tradeoff is that though it's compressed it uses a bit more bandwidth and a bit more CPU. Because of the redundancy to remove chop, it has greater overhead on almost all counts. I have no issue using it, and frequently use it for asterisk asterisk trunking where bandwidth is insufficient for uLaw. Andrew ___ --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] Compatible Asterisk Connectivity Cards : Sangoma
I've never had issue with the Digium cards in testing and as we're looking forward to production systems what compelling reason do I have to pick Sangoma? (I'm not looking for a flame-fest here, but actual compelling reasons, ie Sangoma cards support foo which is needed in situation bar and Digium doesn't support foo/isn't planning to support foo until their next line of cards comes out). Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Monday, April 03, 2006 10:19 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma I wanted to share to everyone the following compatible connectivity products that my company installed in our Asterisk based soft switch. I already sent these to the Asterisk.org site many days ago but for some reason they still have to post it. This feels trollish to me. Please don't feed them. -Brian What intrigued me is the extrapolation from 'we tested on at least 5 servers' to 'works with any commercially available motherboard'! Pete Actually as a point of fact that is a Sangoma guarantee, they guarantee that their cards will work with any PCI-compliant motherboard. MATT--- ___ --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] Dial from php
Flash Operator Panel already has similar functionality, just create a CID entry drag and drop. There may of course be other (better) ways to do this but this is one option/alternative. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant Sent: Friday, March 31, 2006 12:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Dial from php Hi all, Here is the situation. Linux workstation access a web page on a web server (not the one running Asterisk). From that web page, we need to initiate a dial-out on the Asterisk server and once the call is connected, it must ring on the agent's hard phone. Any pointers about how to initiale an Asterisk call from a remove web server? Thanks, Andre Courchesne http://www.net-forces.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 ___ --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 turn key solution
Roughly where are you located? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of mike webb Sent: Friday, March 31, 2006 4:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk turn key solution can anyone recommend a asterisk turn key company. we will need the hardware as well as tech. support 24/7. we'll want all the goodies, voice mail, auto attendant. we have 6 incoming pot lines (all the same number), and 40 normal telephones. we have no interest in changing to ip phones or the pot lines at this time. we're interested in removing our meridian pbx system, installing asterisk, learning how to use it on our own (eventually) ___ --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] RFC Follow Me Find Me script
Just poking this topic as it seems to have been ignored. I still am not
clear as to how/where this script is broken.
If I read this correctly the syntax in column two is the current best
practice for AstDB. It, unless I've missed something below is what I
have used in my script.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew D
Kirch
Sent: Friday, March 10, 2006 11:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RFC Follow Me Find Me script
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
(top posting to follow previous/keep thread sane)
* The applications DBGet and DBPut have been deprecated in favor of
functions. Here is a table of their replacements:
DBGet(foo=family/key)Set(foo=${DB(family/key)})
DBPut(family/key=${foo}) Set(DB(family/key)=${foo})
Johann wrote:
That looks like the dialplan for Asterisk 1.0.x, The AstDB and other
commands have changed in Asterisk 1.2.x(and CVS HEAD). Check the
UPGRADE.txt in the source code directory of Asterisk to get the
details
on all the changes...
--johann
Andrew D Kirch wrote:
This is a follow/find me script that I can't quite get to work,
asterisk wont save forward/${calleridnum} to AstDB... any comments or
thoughts on how to make this work or change it to work differently
are
appreciated. The voice prompts to go with all playback/background
extensions are commented appropriately. I hope this code is of use
to
some of you and any help with a perfected version is of course
appreciated.
[Forward]
exten = s,1,Playback(forward/extension-forwarding)
;Extension Forwarding
exten = s,2,GotoIf($[${CALLERIDNUM}300]?s,5)
;since 1xx is the pattern match for internal extensions anything
less than 300 has to be internal so we already know that that is the
extension they are wanting to forward
exten = s,3,Read(CALLERIDNUM,foward/please-ent-exten,3)
;if it's not have the user enter their 3 digit enternal extension
;please enter the extension you want to forward
exten = s,4,SayNumber(${CALLERIDNUM})
exten = s,5,Background(forward/extension-fwd-menu)
;to hear your current extension forward options press 1, to forward
your phone press 2, to cancel your forwarding press 3
exten = 1,1,Set(FORWARD=${DB(forward/${CALLERIDNUM})})
exten = 1,2,NoOp(FORWARD is ${FORWARD})
exten = 1,3,GotoIf($[${FORWARD}0]?100,3)
exten = 1,4,Playback(forward/your-ext-not-forward)
;your extension is not currently forwarded
exten = 1,5,Goto(Forward,s,5)
;back to main menu
exten = 100,1,Playback(forward/your-ext-forward)
exten = 100,2,SayDigits(${FORWARD})
;your extension is currently forwarded to extension
exten = 100,3,Goto(Forward,s,5)
;back to main menu
exten = 2,1,Read(FORWARD,forward/please-ent-exten)
exten = 2,2,NoOp(FORWARD is ${FORWARD})
exten = 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD}
} )
exten = 2,4,NoOp(forward/${CALLERIDNUM} is ${
DB(forward/${CALLERIDNUM} ) } )
exten = 2,5,Playback(forward/your-ext-forward-saved)
;your extension forward has been saved
exten = 2,6,Goto(Forward,s,5)
exten = 3,1,DBdel(forward/${CALLERIDNUM})
exten = 3,2,PlayBack(forward/exten-forward-cancel)
; your extension forward has been deleted.
exten = 3,3,Goto(Forward,s,1)
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- --
Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org
Security Admin | Summit Open Source Development Group | www.sosdg.org
Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2
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Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
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48W1AzyAkRr3TCgdHwFxIY8=
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[Asterisk-Users] chan_zap ast_pickup_call issue redux
I'm running latest asterisk and zaptel, I have loaded wctdm and lsmod shows that it is in the kernel. I have configured the FXS and FXO ports on my TDM400P, and ztcfg shows both as configured with no errors. When I start asterisk I get the following error: Mar 13 14:07:41 WARNING[10958]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_cal. A search of the web and this mailing list shows issues related to the module not being loaded or zaptel not having been compiled before asterisk. I recompiled asterisk to ensure that it was linked against zaptel and manually deleted the previously installed version of chan_zap.so before doing make install. After following this resolution the issue persists with the same error as before. Any help in getting zaptel working would be greatly appreciated. ___ --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] Planet VoIP Phones
I am attempting to get a planet VIP-150T to register with asterisk 1.2.4. After searching google Ive found what appear to be instructions in German, Russian and Spanish. Has anyone perhaps seen this before? Asterisk is kicking back the following error: Feb 14 09:59:32 NOTICE[21765]: chan_sip.c:10851 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.100.104' - Username/auth name mismatch Sip debug show the following: -- SIP read from 192.168.100.104:5060: REGISTER sip:192.168.100.240:5060 SIP/2.0 From: sip:[EMAIL PROTECTED];tag=c0a86468-13c4-43f1a32d-c2e70-35fb To: sip:[EMAIL PROTECTED] Call-ID: c0a86468-13c4-c-334f-370 CSeq: 14 REGISTER Via: SIP/2.0/UDP 192.168.100.104:5060;branch=z9hG4bK-43f1a32d-c2e75-2618 Max-Forwards: 70 Supported: replaces User-Agent: IP_PHONE (vip150t_050413.bin) Contact: sip:[EMAIL PROTECTED]:5060;expires=90 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.100.104 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.100.104:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.100.104:5060;branch=z9hG4bK-43f1a32d-c2e75-2618;received=192.168.100.104 From: sip:[EMAIL PROTECTED];tag=c0a86468-13c4-43f1a32d-c2e70-35fb To: sip:[EMAIL PROTECTED];tag=as3ec96f19 Call-ID: c0a86468-13c4-c-334f-370 CSeq: 14 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] And below is the corresponding entry from sip.conf: [planet101] type=friend context=internal username=planet101 fromuser=planet101 callerid=Andrew Kirch 101 host=dynamic nat=no canreinvite=yes mailbox=101 disallow=all allow=ulaw ___ --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
