[asterisk-users] Asterisk error message so uncommon, not even Google knows abuot it
I'm setting up a test server with a Digium TE122 and am getting the following error on the console, spewing as fast as it can. Does anyone have any idea what this error might be? [Oct 19 11:24:53] NOTICE[2076]: chan_dahdi.c:3108 my_handle_dchan_exception: PRI got event: Event 59 (59) on D-channel of span 2 -- _ -- 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] Asterisk error message so uncommon, not even Google knows abuot it
On Fri, Oct 19, 2012 at 11:28 AM, Eric Wieling ewiel...@nyigc.com wrote: I'm setting up a test server with a Digium TE122 and am getting the following error on the console, spewing as fast as it can. Does anyone have any idea what this error might be? [Oct 19 11:24:53] NOTICE[2076]: chan_dahdi.c:3108 my_handle_dchan_exception: PRI got event: Event 59 (59) on D-channel of span 2 You have two D channels, why? Some more info would help, like configs and where the PRIs are coming from. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Asterisk error - 1.6.2 SVN - voicemail files corrupted
Hi Tilghman, I am indeed still seeing this issue (emails missing in sequence, and therefore voicemail box not readable), and I have absolutely no third-party vendor solution playing with voicemails. How do I find whether this was a simple bug that was found and fixed in between official versions? (since I am using SVN?) Or how do I debug and find what was the root cause of the issue? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, December 03, 2010 9:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted Hi Tilghman, This particular customer was one of my less sophisticated customer, and I know for sure he isn`t using anything else than Voicemailmain. Not even the basic voicemail to email function. But I will keep an eye opened for any future problem. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, December 03, 2010 4:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted On Thursday 02 December 2010 18:56:44 Mike wrote: 1) How do I fix this? I don't mind manually fixing it when it happens, but what's wrong exactly? There should not be anything within the Asterisk process to cause this. However, I _have_ seen this exact issue with certain 3rd party vendors that supply a tool for checking voicemail via a web interface. The offending tools make no effort to reorder the messages after certain messages are deleted, which is a really bad thing to do. If this is, in fact, the issue, please ask the vendor to fix the interface, because in the current form, it is severely broken behavior. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted
Thanks Jonathan, I did that, it worked. I thought it had something to do with 1.6.2 SVN, since I`ve been using Asterisk for 5 years now and the first time it happened was the day I used SVN. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: Thursday, December 02, 2010 9:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted On Thu, Dec 2, 2010 at 4:56 PM, Mike l...@net-wall.com wrote: Hi, I know I am using SVN, but I was wondering if anybody ever came across this error. There is nothing wrong with using SVN. Well, there isnt a msg.txt file, I can see that. There is a msg0003.txt and msg0005.txt (along with the appropriate wav files). Looking into the directory, all files seem there. Except the sequence doesnt start at . 1) How do I fix this? I dont mind manually fixing it when it happens, but whats wrong exactly? I have seen this once on a 1.6.2 system a while back. I just renamed the TXT and audio files to be sequencial numbers starting at and everything worked again. Asterisk assumes the voicemail message files are named that way, and it errors out if that is not the case. -Jonathan -- _ -- 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] Asterisk error - 1.6.2 SVN - voicemail files corrupted
Hi Tilghman, This particular customer was one of my less sophisticated customer, and I know for sure he isn`t using anything else than Voicemailmain. Not even the basic voicemail to email function. But I will keep an eye opened for any future problem. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, December 03, 2010 4:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted On Thursday 02 December 2010 18:56:44 Mike wrote: 1) How do I fix this? I don't mind manually fixing it when it happens, but what's wrong exactly? There should not be anything within the Asterisk process to cause this. However, I _have_ seen this exact issue with certain 3rd party vendors that supply a tool for checking voicemail via a web interface. The offending tools make no effort to reorder the messages after certain messages are deleted, which is a really bad thing to do. If this is, in fact, the issue, please ask the vendor to fix the interface, because in the current form, it is severely broken behavior. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Asterisk error - 1.6.2 SVN - voicemail files corrupted
Hi, I know I am using SVN, but I was wondering if anybody ever came across this error. I can't read my voicemails because files seems to be corrupted, for lack of a better word. When I do access my messages, I get those errors: [Dec 2 19:45:05] NOTICE[25993]: app_voicemail.c:7432 open_mailbox: Mailbox: /var/spool/asterisk/voicemail/xxx/709/INBOX, expected 0 but found 3 message(s) in box with max threshold of 100. [Dec 2 19:45:05] NOTICE[25993]: app_voicemail.c:7432 open_mailbox: Mailbox: /var/spool/asterisk/voicemail/xxx/709/INBOX, expected 0 but found 3 message(s) in box with max threshold of 100. [snipped] [Dec 2 19:45:07] WARNING[25993]: app_voicemail.c:7207 play_message: No message attribute file?!! (/var/spool/asterisk/voicemail/xxx/709/INBOX/msg.txt) Well, there isn't a msg.txt file, I can see that. There is a msg0003.txt and msg0005.txt (along with the appropriate wav files). Looking into the directory, all files seem there. Except the sequence doesn't start at . I have thousands of mailboxes, only one has been reported as having this problem. There might be more though. 1) How do I fix this? I don't mind manually fixing it when it happens, but what's wrong exactly? 2) If this isn't the right list for this (considering it's a SVN question), what is? I'm using SVN because of the blind transfer issue somebody mentioned yesterday. Mike -- _ -- 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] Asterisk error - 1.6.2 SVN - voicemail files corrupted
On Thu, Dec 2, 2010 at 4:56 PM, Mike l...@net-wall.com wrote: Hi, I know I am using SVN, but I was wondering if anybody ever came across this error. There is nothing wrong with using SVN. Well, there isn’t a msg.txt file, I can see that. There is a msg0003.txt and msg0005.txt (along with the appropriate wav files). Looking into the directory, all files seem there. Except the sequence doesn’t start at . 1) How do I fix this? I don’t mind manually fixing it when it happens, but what’s wrong exactly? I have seen this once on a 1.6.2 system a while back. I just renamed the TXT and audio files to be sequencial numbers starting at and everything worked again. Asterisk assumes the voicemail message files are named that way, and it errors out if that is not the case. -Jonathan -- _ -- 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] Asterisk Error
Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards ___ -- 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 Error
This means that no ethernet interface is found for seeding the global EID. So you will have to set it manually. :) Pretty clear. On Thu, Jul 16, 2009 at 11:08 PM, michel freihamich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards ___ -- 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 Error
I would guess that the MAC address of an Ethernet adaptor is used as a seed for a pseudorandom number generation algorithm that is used to create a GUID (Globally Unique Identifier) for your DUNDI node. But that requires an Ethernet adaptor. Ali Jawad wrote: This means that no ethernet interface is found for seeding the global EID. So you will have to set it manually. :) Pretty clear. On Thu, Jul 16, 2009 at 11:08 PM, michel freihamich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards ___ -- 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 -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- 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 Error
On Fri, Jul 17, 2009 at 2:08 AM, michel freiha mich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards Add: noload = dundi To your modules.conf. That should fix it. Do you want to use dundi? What does ifconfig say? I assume you have a NIC? Is it up and all that when you start Asterisk? Have you tried downing it, setting all the variables (maybe even the MAC to be thorough) and then bringing it back up before starting Asterisk? Otherwise what kind of NIC? Do you have an old 3Com laying around you can pop in it? Open a bug report? -- 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
Re: [asterisk-users] Asterisk Error
Dear Sir I did what you asked me to do...i added the following to /etc/opt/asterisk/modules.conf noload = dundi -bash-3.00# ifconfig -a lo0: flags=2001000849UP,LOOPBACK,RUNNING,MULTICAST,IPv4,VIRTUAL mtu 8232 index 1 inet 127.0.0.1 netmask ff00 eri0: flags=1000843UP,BROADCAST,RUNNING,MULTICAST,IPv4 mtu 1500 index 2 inet 192.168.0.178 netmask ff00 broadcast 192.168.0.255 ether 0:3:ba:f2:d2:ea Yes I have a NIC, Up and running and I can SSH the server from that NIC Regards On Fri, Jul 17, 2009 at 3:21 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: On Fri, Jul 17, 2009 at 2:08 AM, michel freiha mich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards Add: noload = dundi To your modules.conf. That should fix it. Do you want to use dundi? What does ifconfig say? I assume you have a NIC? Is it up and all that when you start Asterisk? Have you tried downing it, setting all the variables (maybe even the MAC to be thorough) and then bringing it back up before starting Asterisk? Otherwise what kind of NIC? Do you have an old 3Com laying around you can pop in it? Open a bug report? -- 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 Error
It may be** noload = pbx_dundi.so or some such. Sorry for being so vague in my original answer but googling noload dundi would have given you the same answer I just did. You could probably safely just delete pbx_dundi.so instead/as well or recompile Asterisk, do a make menuselect and remove dundi then make make install. That should at least solve your dundi issue. Thanks, Steve Totaro On Fri, Jul 17, 2009 at 9:01 AM, michel freiha mich...@gmail.com wrote: Dear Sir I did what you asked me to do...i added the following to /etc/opt/asterisk/modules.conf noload = dundi -bash-3.00# ifconfig -a lo0: flags=2001000849UP,LOOPBACK,RUNNING,MULTICAST,IPv4,VIRTUAL mtu 8232 index 1 inet 127.0.0.1 netmask ff00 eri0: flags=1000843UP,BROADCAST,RUNNING,MULTICAST,IPv4 mtu 1500 index 2 inet 192.168.0.178 netmask ff00 broadcast 192.168.0.255 ether 0:3:ba:f2:d2:ea Yes I have a NIC, Up and running and I can SSH the server from that NIC Regards On Fri, Jul 17, 2009 at 3:21 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: On Fri, Jul 17, 2009 at 2:08 AM, michel freiha mich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards Add: noload = dundi To your modules.conf. That should fix it. Do you want to use dundi? What does ifconfig say? I assume you have a NIC? Is it up and all that when you start Asterisk? Have you tried downing it, setting all the variables (maybe even the MAC to be thorough) and then bringing it back up before starting Asterisk? Otherwise what kind of NIC? Do you have an old 3Com laying around you can pop in it? Open a bug report? -- 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 -- 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
Re: [asterisk-users] Asterisk Error
On Fri, 17 Jul 2009, Steve Totaro wrote: It may be** noload = pbx_dundi.so or some such. Sorry for being so vague in my original answer but googling noload dundi would have given you the same answer I just did. Oh come on Steve, you should have known you would end up googling when the OP starts with a great subject like Asterisk Error. At least they didn't misspell Asterisk or use the ever so searchable * I'm expecting the list to atrophy (Idiocracy anyone?) to the point every post will carry the subject *? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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: error while loading shared libraries: libiksemel.so.3: cannot open shared object file: No such file or directory
How I can disable Gtalk Jabber module?Thanks# asterisk -vcasterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared object file: No such file or directory===Centos4.4 2.6.9-34.0.2.ELzaptel 1.4.1asterisk 1.4.2iksemel 1.2Dmitri Smirnoff msn: [EMAIL PROTECTED]: 613 693 1299 ext 120 ___ --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: error while loading shared libraries: libiksemel.
From: Dmitri Smirnoff [EMAIL PROTECTED] Date: Sat, 24 Mar 2007 21:11:17 -0400 How I can disable Gtalk Jabber module?Thanks# asterisk -vcasterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared objectfile: No such file or directory===Centos4.4 2.6.9-34.0.2.ELzaptel 1.4.1asterisk 1.4.2iksemel 1.2Dmitri Smirnoff msn: [EMAIL PROTECTED]: 613 693 1299 ext 120 Rerun make menuselect? Yuan Liu ___ --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 error
I usually see this when doing operations on variables that are blank. In your case, the input is ' + 1'. Clearly there was something to the left of the + but it's blank. If you're adding one to something, make sure there is a number on the left side of the plus sign. Probably by initializing the variable to zero before you first use it. /edg --On Monday, February 20, 2006 1:56 PM -0300 Dov Bigio [EMAIL PROTECTED] wrote: Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: + 1 ^ 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source. Any ideas? a Thank you Dov ___ --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 error
Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:+ 1^2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source. Any ideas? a Thank you Dov ___ --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 error
Dov Bigio wrote: Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: What part of your dial plan is generating the error? Can you post it? Doug ___ --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 error
Dov Bigio wrote: Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: What part of your dial plan is generating the error? Can you post it? Doug This sounds a lot like the error Doug was getting when he tried to increment a variable before it actually was defined. Please post the part of the dialplan that causes this error and we will probably be able to figure it out. -MC ___ --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
