Re: [asterisk-users] Yellow or Red alarm on TE110P ????
the cable is a simple cable break or: the cable schema we see bellow (1-4 2-5): 1---\ /---1 2-\ / \ /-2 3-/-\-/-\-3 4__/ / \ \4 5__ / \_5 6-6 7-7 8-8 kind regards ___ --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] AJAM..is a BUG?
hi guys i have created a plugin for jquery for asterisk ajax interfacement. the interfacement work with ajam and on firefox work very well, the problem is with IE :-( an example: the url is: asterisk/mxml i want login on manager system and the string command is: action=loginusername=myusernamesecret=mysecret I have tested with firefoz and i receive the correct XML response, the jquery plugin work perfect, but on IE i receive an error message from Asterisk Server 501 Error, unsopported method. i have opened the link directly with IE and i receive the XML response, but when i contact the AJAM web server via AJAX i receive one error. have any idea? this is a BUG? thanks for your reply p.s. I have resolved my first problem with sellvoip ITSP, thanks to all ___ --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] Error Message.
Hello, i just installed asterisk 1.2.15. I got this error message. Somebody can help me? Thank You. Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled. Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application 'Pickup' Feb 27 11:47:44 WARNING[17086] loader.c: app_directed_pickup.so: load_module failed, returning -1 Feb 27 11:47:44 WARNING[17086] loader.c: Loading module app_directed_pickup.so failed! ___ --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] Cisco 7960
Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --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] Error Message.
On Tue, 2007-02-27 at 11:52 +0200, Jonson Player wrote: Hello, i just installed asterisk 1.2.15. I got this error message. Somebody can help me? Thank You. Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled. Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application 'Pickup' Feb 27 11:47:44 WARNING[17086] loader.c: app_directed_pickup.so: load_module failed, returning -1 Feb 27 11:47:44 WARNING[17086] loader.c: Loading module app_directed_pickup.so failed! Where is there an error message in the above? app_directed_pickup.so is just saying some other version of pickup is already in place. Need much more information. -- Dave Cotton [EMAIL PROTECTED] ___ --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] AJAM..is a BUG?
[EMAIL PROTECTED] wrote: hi guys i have created a plugin for jquery for asterisk ajax interfacement. the interfacement work with ajam and on firefox work very well, the problem is with IE :-( an example: the url is: asterisk/mxml i want login on manager system and the string command is: action=loginusername=myusernamesecret=mysecret I have tested with firefoz and i receive the correct XML response, the jquery plugin work perfect, but on IE i receive an error message from Asterisk Server 501 Error, unsopported method. You probably need to do a GET, not HEAD, POST, PUT or something. i have opened the link directly with IE and i receive the XML response, but when i contact the AJAM web server via AJAX i receive one error. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mgcp codec problem about ulaw
Hi: I have a mgcp.conf and a mgcp_additional.conf which records the special information about the extensions. And i found if i use ulaw in the general context in mgcp.conf,then all the registered extensions can make both outbound and inbound calls,the mgcp.conf is following: [general] port = 2727 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw; can be disable and do no effect #include mgcp_additional.conf But if i disable ulaw and the mgcp.conf is following: [general] port = 2727 bindaddr = 0.0.0.0 disallow=all ;allow=ulaw ; be disable and all the extensions can not be called allow=alaw; can be disable and do no effect #include mgcp_additional.conf then all the registered extensions can make outbound calls but can not be called by sip phone.The output of asterisk is following: asterisk1*CLI -- Executing NoOp(SIP/1000-084d94b8, here) in new stack -- Executing Dial(SIP/1000-084d94b8, MGCP/[EMAIL PROTECTED]|45) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/1000-084d94b8, ) in new stack == Spawn extension (from-internal, 6000, 3) exited non-zero on 'SIP/1000-084d94b8' -- Executing Macro(SIP/1000-084d94b8, hangupcall) in new stack -- Executing ResetCDR(SIP/1000-084d94b8, w) in new stack -- Executing NoCDR(SIP/1000-084d94b8, ) in new stack -- Executing GotoIf(SIP/1000-084d94b8, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf(SIP/1000-084d94b8, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait(SIP/1000-084d94b8, 5) in new stack -- Executing Hangup(SIP/1000-084d94b8, ) in new stack == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/1000-084d94b8' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/1000-084d94b8' the file mgcp_additional.conf is following: [6000] accountcode = 6000 context = [ext-local] callerid = 6000 6000 host = dynamic disallow = all allow = g723.1 allow = alaw allow = g729 allow = ulaw dtmfmode = rfc2833 nat = no line = 6000 the file extensions_additional.conf is following: [ext-local] exten = 6000,1,NoOp(here) exten = 6000,2,Dial(MGCP/[EMAIL PROTECTED],45) exten = 6000,3,Hangup exten = 6000,hint,MGCP/6000 Thanks in advance. ___ --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] Cisco 7960
can you give a bit more info? I know that you need nat=never for example - Original Message - From: Khaled To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Cisco 7960
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RE: [asterisk-users] Cisco 7960
I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From: Khaled mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --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] CWI, call-limit and incominglimit
On 2/24/07, Pavel Jezek [EMAIL PROTECTED] wrote: Brian Capouch wrote: But the included comments say, The user part of a type=friend call will still be affected by the call limit Those seem to be in conflict, but perhaps it's just my parser :-) Could someone clueful explain? I interpret this that asterisk _internally_ still counting calls for both user and peer, but actually limits calls only for peers... :-\ Thanks for all of the pointers on this - I think merging the limitonpeers change from trunk into 1.2.15 is my favourite option right now. Or should I just take chan_sip.c from trunk? Would that be fairly safe? I think that the problem I was having is actually related to sip reload not clearing down an old call-limit setting to its default if the option is removed from a sip user/peer (as opposed to setting call-limit: 0 to disable it.) I have not confirmed this to the point where I can open a bug, but someone might want to check that out? :) Also, from reading the code... Is it worth updating sip show inuse to reflect the setting of limitonpeers ? Regards, Steve ___ --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] CWI, call-limit and incominglimit
On 2/27/07, Steve Davies [EMAIL PROTECTED] wrote: Thanks for all of the pointers on this - I think merging the limitonpeers change from trunk into 1.2.15 is my favourite option right now. Or should I just take chan_sip.c from trunk? Would that be fairly safe? Err... What I meant was shall I take chan_sip.c from the head of the 1.2 branch, but now I see that the limitonpeers option is a 1.4 feature. Steve ___ --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] How to get values of local channels context
Hello.
Take a look about function SIPPEER (asterisk -rx show function
SIPPEER).
It helps how to use peer information.
Regards.
José Luis
El lun, 26-02-2007 a las 23:09 -0800, Yuan LIU escribió:
From: kjcsb [EMAIL PROTECTED]
Date: Mon, 26 Feb 2007 22:32:29 -0800 (PST)
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Context
macro-test) in new stack
I want to get 116-2000 somehow.
Any suggestions would be appreciated.
So use ${MACRO_CONTEXT} .
Thanks
But doesn't this give the calling context which, if itself is another
macro, will still not give me what I want? If macro-test is called by
macro-first then ${MACRO_CONTEXT} = macro-first. Surely there's a way to
get the context directly from the Local channel itself?
Cameron
If nested macro calls are necessary, define an inheritable local variable,
e.g., __real-context. Two _'s enables infinite inheritance. Hope this
helps.
Yuan Liu
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[asterisk-users] Autentication
Hi, i have a doubt about autentication in asterisk. it's possible to integration the asterisk with the other server for autentication, for example kerberos, ou other? i want to implement asterisk in a department of university, but it's necessary autentication by students, login and password for example. thanks. -- Carlos Jerónimo ___ --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] Cisco 7960
Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks --- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled To:'Asterisk Users Mailing List - Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 - * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -- Thx MAG ___ --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] Forbidden - wrong password on authentication for INVITE
Hi, I'm using messagenet VoIP provider, I can make calls but I cannot receive calls. When I call my VoIP number the phone rings but when I pick up the call drops and I get this message on Asterisk console: *Forbidden - wrong password on authentication for INVITE to ...* Is there anybody who knows why? Giorgio Incantalupo ___ --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
FW: [asterisk-users] Cisco 7960
Softphone Eyebeam v 1.5.2 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A. Gombolaty Sent: Tuesday, February 27, 2007 2:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled mailto:[EMAIL PROTECTED] To:'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * _ ___ --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 -- Thx MAG * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the
AW: [asterisk-users] Cisco 7960
Hi Carlos, Check out Asterisk LDAP authentication: http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP Greetz, [EMAIL PROTECTED] _ Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Mohamed A. Gombolaty Gesendet: 27 February 2007 13:03 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Cisco 7960 Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled mailto:[EMAIL PROTECTED] To:'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * _ ___ --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 -- Thx MAG ___ --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: FW: [asterisk-users] Cisco 7960
Dear All, Please send the sip configuration for both phones along with a debug from asterisk when you try to call from cisco to the eyebeam? also are you trying to make them call peer to peer or not? What I am suspecting is that there must be something mismatching when the cisco phone tries to call the softphone you just need to focus on the debug and check the configuration. Thx MAG Khaled wrote: Softphone Eyebeam v 1.5.2 --- From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A. Gombolaty Sent: Tuesday, February 27, 2007 2:03 PM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks - From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To:Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled To:'Asterisk Users Mailing List - Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of
[asterisk-users] OutBound Proxy calls failing
Hello Users, Good AfterNoon to all I'm Mainly focused on OpenSER and Asterisk Integration. I didn't Find any solution of My Question ? Till now I'm doing only communicating OpenSER and Asterisk through SIP Channel only. User in Asterisk can Call to OpenSER and also vice-versa . But My Question ? I have one VoIP Service line from Voyage ( SIP change ), I want, if one of the User in Asterisk has to call Voyage service, and Call from Voyage line to has to Asterisk server, that play the IVR. -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [asterisk-users] AJAM..is a BUG?
You probably need to do a GET, not HEAD, POST, PUT or something. The method is GET and with Firefox all work well i dont understand? ___ --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] Authentication Command
Anyone else experiencing a slow authentication command. I noticed this command takes about 1.5 - 2 seconds of silence before it asked for password, then another 2 sec of silence before it moves froward after that. Any ideas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AJAM..is a BUG?
[EMAIL PROTECTED] wrote: You probably need to do a GET, not HEAD, POST, PUT or something. The method is GET and with Firefox all work well i dont understand? Somehow IE seems to send a different request. I'm not familiar with jquery nor do I use IE so I can't tell if jquery or IE is to blame. Did you compare the traces of the requests a) when you enter the URL into the address line (works) b) when you use jquery (does not work) tcpflow -c tcp port 5038 or tcpdump -s 0 -A tcp port 5038 Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AJAM..is a BUG?
Philipp Kempgen wrote: tcpflow -c tcp port 5038 s/5038/8088/ :-) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VLAN vs RealLan
Given a choice, and a green-field site, would you a) Have a separate network (switches etc) for your data and phone b) Use the same network, but use VLAN's ?? What are the pro's and con's of each ? TIA Julian ___ --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] NetFilter (IPTables)
I have this running on my Asterisk server, and have opened up ports UDP/5060 and UDP/1-2 but for some reason when I try and connect too my SIP extension it does not work. Are these the correct ports ? -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get values of local channels context
Lee Jenkins wrote:
kjcsb wrote:
The variable ${CONTEXT} stores the value of the current context.
However if we are in a macro that will be the name of the macro. How
do I access the name of the local channel's context.
For example:
[macro-test]
exten = s,n,NoOp(Context ${CONTEXT})
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2
mailto:Local/[EMAIL PROTECTED],2, Context macro-test) in new
stack
I want to get 116-2000 somehow.
Hi,
Check out ${MACRO_CONTEXT}
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro
Check out /path/to/src/asterisk/doc/README.variables
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Re: [asterisk-users] NetFilter (IPTables)
Hi, I've found this doc helpful in configuring my iptables: http://www.voip-info.org/wiki-Asterisk+firewall+rules Following those settings, my devices register and function properly. Alex On 2/27/07, -- [ UxBoD ] -- [EMAIL PROTECTED] wrote: I have this running on my Asterisk server, and have opened up ports UDP/5060 and UDP/1-2 but for some reason when I try and connect too my SIP extension it does not work. Are these the correct ports ? -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --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] RES: asterisk-users Digest, Vol 31, Issue 115
Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled To:'Asterisk Users Mailing List - Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 - * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -- Thx MAG -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070227/69ec15 d6/attachment-0001.htm -- Message: 4 Date: Tue, 27 Feb 2007 13:07:50 +0100 From: Giorgio Incantalupo [EMAIL PROTECTED] Subject: [asterisk-users] Forbidden - wrong password on authentication for INVITE To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I'm using messagenet VoIP provider, I can make calls but I cannot receive calls. When I call my VoIP number the phone rings but when I pick up the call drops and I get this message on Asterisk console: *Forbidden - wrong password on authentication for INVITE to ...* Is there anybody who knows why? Giorgio Incantalupo -- Message: 5 Date: Tue, 27 Feb
Re: [asterisk-users] RES: asterisk-users Digest, Vol 31, Issue 115
I've had a stable call center running since 2004, with only occasional
maintenance required. The only time it ever crashes is when I let it
fill up it's disks with call recordings.
I've had another system up and running since 2003 that hasn't hardly had
anything done to it.
Both of these systems are running the Stable branch of Asterisk on
Compaq (HP) servers and Red Hat Enterprise Server.
As with any system, I would recommend that you create a test environment
and test all of your patches, changes, etc. before applying them to your
Production system(s).
Good luck and have fun!
Roberto wrote:
Questions:
Does anyone have a really STABLE asterisk system running about one year
without need to restart the service or the SERVER ?
Does anyone have a production Call Centre saled that don't lockup and is
stable for 6 months ?
I'm asking this questions because we have choose Asterisk for our call
centre solution but, since the bugtracker only grows and people still want
to stuck more and more features without solve CRITICAL and crash bugs.
Can someone answer my questions?
Roberto
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de
[EMAIL PROTECTED]
Enviada em: terça-feira, 27 de fevereiro de 2007 09:26
Para: asterisk-users@lists.digium.com
Assunto: asterisk-users Digest, Vol 31, Issue 115
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When replying, please edit your Subject line so it is more specific
than Re: Contents of asterisk-users digest...
Today's Topics:
1. Re: How to get values of local channels context (Jos? Luis G?mez)
2. Autentication ( Carlos Jer?nimo )
3. Re: Cisco 7960 (Mohamed A. Gombolaty)
4. Forbidden - wrong password on authentication for INVITE
(Giorgio Incantalupo)
5. FW: [asterisk-users] Cisco 7960 (Khaled)
6. AW: [asterisk-users] Cisco 7960 (Roland Ndaka Fru)
7. Re: FW: [asterisk-users] Cisco 7960 (Mohamed A. Gombolaty)
--
Message: 1
Date: Tue, 27 Feb 2007 08:50:58 -0300
From: Jos? Luis G?mez [EMAIL PROTECTED]
Subject: Re: [asterisk-users] How to get values of local channels
context
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Hello.
Take a look about function SIPPEER (asterisk -rx show function
SIPPEER).
It helps how to use peer information.
Regards.
Josi Luis
El lun, 26-02-2007 a las 23:09 -0800, Yuan LIU escribis:
From: kjcsb [EMAIL PROTECTED]
Date: Mon, 26 Feb 2007 22:32:29 -0800 (PST)
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Context
macro-test) in new stack
I want to get 116-2000 somehow.
Any suggestions would be appreciated.
So use ${MACRO_CONTEXT} .
Thanks
But doesn't this give the calling context which, if itself is another
macro, will still not give me what I want? If macro-test is called by
macro-first then ${MACRO_CONTEXT} = macro-first. Surely there's a way to
get the context directly from the Local channel itself?
Cameron
If nested macro calls are necessary, define an inheritable local variable,
e.g., __real-context. Two _'s enables infinite inheritance. Hope this
helps.
Yuan Liu
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--
Message: 2
Date: Tue, 27 Feb 2007 12:02:19 +
From: Carlos Jer?nimo [EMAIL PROTECTED]
Subject: [asterisk-users] Autentication
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi, i have a doubt about autentication in asterisk.
it's possible to integration the asterisk with the other server for
autentication, for example kerberos, ou other?
i want to implement asterisk in a department of university, but it's
necessary autentication by students, login and password for example.
thanks.
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Re: [asterisk-users] VLAN vs RealLan
Julian Lyndon-Smith wrote: Given a choice, and a green-field site, would you a) Have a separate network (switches etc) for your data and phone b) Use the same network, but use VLAN's ?? What are the pro's and con's of each ? TIA Julian ___ --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 We use a hybrid approach. The two methods can live happily together by trunking the VLANs to all switches but allocating particular switches to voice and others to data. One VLAN is dedicated to VoIP. The PoE capable switches are generally dedicated to VoIP devices but some ports are assigned to other VLANs to connect relatively low traffic devices that require PoE like wireless access points and security cameras. Occasionally a VoIP device may be connected to a data switch due to cabling or other issues. Pro's:- Reduced cost, greater flexibility, best of all worlds really. Data traffic will not consume switching or wire bandwidth required for VoIP as the traffic is (mostly) on separate switches. Data switches get basic UPS support ( 30 mins, like the servers). VoIP switches get full UPS support (2 hrs). This minimizes the cost of battery backup while providing the expected duration for phones. Fewer switches required than physically separate networks We can connect devices to any switch if and when required. Con's:- More complexity to manage. You do need to understand ethernet and traffic issues. Mixed traffic trunks have to be carefully managed to preserve VoIP QoS. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RES: asterisk-users Digest, Vol 31, Issue 115
config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks --- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled To:'Asterisk Users Mailing List - Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 - * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -- Thx MAG -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070227/69ec15 d6/attachment-0001.htm -- Message: 4 Date: Tue, 27 Feb 2007 13:07:50 +0100 From: Giorgio Incantalupo [EMAIL PROTECTED] Subject: [asterisk-users] Forbidden - wrong password on authentication for INVITE To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID
[asterisk-users] Do I understand GROUPs correctly?
Hi,
I was under the impression that Set(GROUP()=1234) incremented some value
associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten = s,1,Set(GROUP()=1234)
exten = s,n,Set(GROUP()=1234)
exten = s,n,Noop(Used channels: ${GROUP_COUNT(1234})
I get this in the CLI:
-- Executing Set(IAX2/test-2, GROUP()=1234) in new stack
-- Executing Set(IAX2/test-2, GROUP()=1234) in new stack
-- Executing NoOp(IAX2/test-2, Used channels: 1) in new stack
I'm trying to limit the amount of channel used by some customers, but the
incrementation doesn't seem to work. I'm using Asterisk 1.2.13.
My real life example (the above is clearly a proof of concept that is
failing) is that someone calling an IVR uses one channel. Someone calling
the IVR and being eventually transfered to a cell phone
(PSTN-ASTERISKPSTN) uses two channels. Unfortunately the number
doesn't increased as planned, it stays at 1 channel used.
Regards,
Mike
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Re: [asterisk-users] Do I understand GROUPs correctly?
Mike wrote:
Hi,
I was under the impression that Set(GROUP()=1234) incremented some
value associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten = s,1,Set(GROUP()=1234)
exten = s,n,Set(GROUP()=1234)
exten = s,n,Noop(Used channels: ${GROUP_COUNT(1234})
If this is a direct copy/paste then your error is in line 3. You have a
} positioned incorrectly. My example below:
exten = _35XX,1,Set(GROUP()=Max_Calls)
exten = _35XX,n,NoOP(Active Calls: ${GROUP_COUNT(Max_Calls)})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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[asterisk-users] call-limit in 1.2 HEAD
Hi, Could someone double-check a behaviour I am seeing in 1.2 SVN HEAD In sip.conf, create a type=friend entry with call-limit=1 1) Place an outbound call from the device 2) Place a call in to the device sip show inuse is now something like: * User name In use Limit x-lite1 1 * Peer name In use Limit x-lite1 1 If you then hangup (from the device) first the outbound call, and then the inbound call, I get a lost refcount: * User name In use Limit x-lite0 1 * Peer name In use Limit x-lite1 1 though there are no calls in progress anymore. From the debug output, it seems that the hang-up which should decrement the peer-inUse count tries to decrement the user-inUse count instead. Strangely, if I hangup the calls in reverse order, inbound and then outbound, everything is happy, and if the remote party (not the call-limited device) hangs up, then it also seems to be happy. This is all quite reproducible! I notice that in 1.2.15 the behaviour is different in that all of the inUse counting is done against the user-inUse counter anyway. Thanks, Steve ___ --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] Billing Telephone Number (BTN)
I have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --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] Do I understand GROUPs correctly?
Actually it wasnt a straight paste. The straight cut and paste is:
exten = s,1,Set(GROUP()=${VAR})
exten = s,n,Set(GROUP()=${VAR})
exten = s,n,Noop(Used channels: ${GROUP_COUNT(${VAR})})
I believe that's good. But The group count is not 2, but 1. I thought
I'd be 2 since I called Set(group) twice.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, February 27, 2007 10:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Do I understand GROUPs correctly?
Mike wrote:
Hi,
I was under the impression that Set(GROUP()=1234) incremented some
value associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten = s,1,Set(GROUP()=1234)
exten = s,n,Set(GROUP()=1234)
exten = s,n,Noop(Used channels: ${GROUP_COUNT(1234})
If this is a direct copy/paste then your error is in line 3. You have a }
positioned incorrectly. My example below:
exten = _35XX,1,Set(GROUP()=Max_Calls)
exten = _35XX,n,NoOP(Active Calls: ${GROUP_COUNT(Max_Calls)})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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[asterisk-users] Grandstream SYSLOG error codes
Hello, I've enabled BT-200's SYSLOG logging, and I get some message whose meaning is obscure to me. In particular, in a day I got the Deletion of invalid timer message almost ten times from one phone, which has some call problems. Can someone point me to a resource on BT200 error codes? Thanks, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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] Do I understand GROUPs correctly?
Doug Lytle wrote:
Mike wrote:
Hi,
I was under the impression that Set(GROUP()=1234) incremented some
value associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten = s,1,Set(GROUP()=1234)
exten = s,n,Set(GROUP()=1234)
exten = s,n,Noop(Used channels: ${GROUP_COUNT(1234})
If this is a direct copy/paste then your error is in line 3. You have a
} positioned incorrectly. My example below:
exten = _35XX,1,Set(GROUP()=Max_Calls)
exten = _35XX,n,NoOP(Active Calls: ${GROUP_COUNT(Max_Calls)})
Apart from that you assign the group 1234 twice to the *same*
channel. So GROUP_COUNT(1234) correctly reports only *1*
channel to be in that group.
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] Do I understand GROUPs correctly?
Mike wrote:
Actually it wasn’t a straight paste. The straight cut and paste is:
exten = s,1,Set(GROUP()=${VAR})
exten = s,n,Set(GROUP()=${VAR})
exten = s,n,Noop(Used channels: ${GROUP_COUNT(${VAR})})
I've never tried using variables with GROUP(), but am guessing it's
permitted.
Try adding another Set and see if the count moves to 2. It may be
starting at 0?
Doug
--
Ben Franklin quote:
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deserve neither Liberty nor Safety.
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Re: [asterisk-users] Do I understand GROUPs correctly?
Philipp Kempgen wrote: Doug Lytle wrote: Apart from that you assign the group 1234 twice to the *same* channel. So GROUP_COUNT(1234) correctly reports only *1* That would be it! 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
RE: [asterisk-users] Do I understand GROUPs correctly?
Ok, that sort of makes sense. But what I am doing is passing off a call
into my Asterisk system to a cell phone. I want this to count as 2
channels. So, I am doing, in effect, this kind of algo:
Answer the call
Set(Group) to increment channel to 1
Play IVR, go into menus, etc.
Eventually go into a Set(group) again to increment channel before dialing a
cell phone using a dial(cellphone#) cmd.
If that doesn't work, how do I accomplish the same kind of thing elegantly?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: Tuesday, February 27, 2007 10:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Do I understand GROUPs correctly?
Doug Lytle wrote:
Mike wrote:
Hi,
I was under the impression that Set(GROUP()=1234) incremented some
value associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten = s,1,Set(GROUP()=1234)
exten = s,n,Set(GROUP()=1234)
exten = s,n,Noop(Used channels: ${GROUP_COUNT(1234})
If this is a direct copy/paste then your error is in line 3. You have
a } positioned incorrectly. My example below:
exten = _35XX,1,Set(GROUP()=Max_Calls) exten = _35XX,n,NoOP(Active
Calls: ${GROUP_COUNT(Max_Calls)})
Apart from that you assign the group 1234 twice to the *same* channel. So
GROUP_COUNT(1234) correctly reports only *1* channel to be in that group.
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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RE: [asterisk-users] Do I understand GROUPs correctly?
Greetings Mike, On Tue, 2007-02-27 at 11:28 -0500, Mike wrote: Ok, that sort of makes sense. But what I am doing is passing off a call into my Asterisk system to a cell phone. I want this to count as 2 channels. So, I am doing, in effect, this kind of algo: Answer the call Set(Group) to increment channel to 1 Play IVR, go into menus, etc. Eventually go into a Set(group) again to increment channel before dialing a cell phone using a dial(cellphone#) cmd. If that doesn't work, how do I accomplish the same kind of thing elegantly? From show application Dial: If the OUTBOUND_GROUP variable is set, all peer channels created by this application will be put into that group (as in Set(GROUP()=...). This would make it so that your outgoing channel would be in the group and the count would be 2. Is this what you are looking for? Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf limitonpeers=yes in asterisk 1.4
Hi, An observation on this feature, which I may have completely misunderstood, so flame away if I am being dumb :) Looking at the code, setting limitonpeers=yes causes all user and peer calls to be ref-counted as if they are peer calls (assuming a user and peer of the same name exist). A side-effect of this is that an incoming call seems to have its call-limit evaluated based on the peer's, rather than the user's settimg, unless no call-limit has been set against the user, in which case the peer's call-limit is ignored too. I also noticed that if an inbound (user) call is blocked based on the above, then unref_peer(p) is called, instead of unref_user(u) - I have no clue what that does, so it may be quite safe :) Regards, Steve ___ --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] Do I understand GROUPs correctly?
Dear Mike,
I had wanted to do something that is similar to your need as I wanted to be
able to add one active channel in multiple groups, it worked with The Ramon's
example in the link below which uses categories beside the set command, note
there are two examles depending on the asterisk version you are using:
http://www.voip-info.org/wiki/view/asterisk+cmd+setgroup
Thx
MAG
Mike wrote:
Ok, that sort of makes sense. But what I am doing is passing off a call
into my Asterisk system to a cell phone. I want this to count as 2
channels. So, I am doing, in effect, this kind of algo:
Answer the call
Set(Group) to increment channel to 1
Play IVR, go into menus, etc.
Eventually go into a Set(group) again to increment channel before dialing a
cell phone using a dial(cellphone#) cmd.
If that doesn't work, how do I accomplish the same kind of thing elegantly?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: Tuesday, February 27, 2007 10:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Do I understand GROUPs correctly?
Doug Lytle wrote:
Mike wrote:
Hi,
I was under the impression that Set(GROUP()=1234) incremented some
value associated with 1234.
So if I did the same thing twice, I'd get a group count of 2.
Ex:
exten = s,1,Set(GROUP()=1234)
exten = s,n,Set(GROUP()=1234)
exten = s,n,Noop(Used channels: ${GROUP_COUNT(1234})
If this is a direct copy/paste then your error is in line 3. You have
a } positioned incorrectly. My example below:
exten = _35XX,1,Set(GROUP()=Max_Calls) exten = _35XX,n,NoOP(Active
Calls: ${GROUP_COUNT(Max_Calls)})
Apart from that you assign the group 1234 twice to the *same* channel. So
GROUP_COUNT(1234) correctly reports only *1* channel to be in that group.
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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--
Thx
MAG
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[asterisk-users] running asterisk through cellphone
hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via data-cable (or bluetooth?) and use this as the single line to call-in. searching the asterisk-forums I found mentions of chan_cellphone, which is probably a patch for exactly this kind of usage, right? I'ld be thankful if you could just point me to the right direction (I'm quite new to asterisk). thx in advance! 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
[asterisk-users] Polycom Firmware
Hi Guys, A while back (several months ago) I was having issues with wmy Polycom's and Asterisk. I was told to use a specific set of firmware and sip version. I am unable to find that email. Anyone know which ones work well with Asterisk ? (I believe it was 2.x ) Thanks, Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Firmware
Dovid B wrote: Hi Guys, A while back (several months ago) I was having issues with wmy Polycom's and Asterisk. I was told to use a specific set of firmware and sip version. I am unable to find that email. Anyone know which ones work well with Asterisk ? (I believe it was 2.x ) I have yet to move away from 1.5.2 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
Re: [asterisk-users] running asterisk through cellphone
What is the cellular connection for ? Are you using this for inbound or the clients will call in in from thier cell phones ? If you need incoming (and or ourgoing) lines you can get one from an ITSP. If you want to use your cell phone you can use chan_cellphone. In order to use it you will need to install the patch. For more information have at look at this: http://bugs.digium.com/view.php?id=8919 - Original Message - From: Michael Kamleitner To: asterisk-users@lists.digium.com Sent: Tuesday, February 27, 2007 6:54 PM Subject: [asterisk-users] running asterisk through cellphone hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via data-cable (or bluetooth?) and use this as the single line to call-in. searching the asterisk-forums I found mentions of chan_cellphone, which is probably a patch for exactly this kind of usage, right? I'ld be thankful if you could just point me to the right direction (I'm quite new to asterisk). thx in advance! 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Firmware
- Original Message - From: Doug Lytle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 27, 2007 7:05 PM Subject: Re: [asterisk-users] Polycom Firmware Dovid B wrote: Hi Guys, A while back (several months ago) I was having issues with wmy Polycom's and Asterisk. I was told to use a specific set of firmware and sip version. I am unable to find that email. Anyone know which ones work well with Asterisk ? (I believe it was 2.x ) I have yet to move away from 1.5.2 Doug Doug is this for the sip version or firmware ? As far as I know once you go beyond a certain firmware version with polycom you cant go back. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Firmware
On Tue, 2007-02-27 at 19:14 +0200, Dovid B wrote: Doug is this for the sip version or firmware ? As far as I know once you go beyond a certain firmware version with polycom you cant go back. Dovid We used bootrom version 2.6.1. And yes, once you go to version 3.x, you cannot go back. Found that phones purchased recently have been shipping with 3.1.2 and higher... ___ --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] rtc: lost some interrupts at 1024Hz
Hi im having this message in my console and dmesg. rtc: lost some interrupts at 1024Hz im not sure what this is. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Firmware
Dovid B wrote: - Original Message - From: Doug Lytle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 27, 2007 7:05 PM Subject: Re: [asterisk-users] Polycom Firmware Dovid B wrote: Hi Guys, A while back (several months ago) I was having issues with wmy Polycom's and Asterisk. I was told to use a specific set of firmware and sip version. I am unable to find that email. Anyone know which ones work well with Asterisk ? (I believe it was 2.x ) I have yet to move away from 1.5.2 Doug Doug is this for the sip version or firmware ? As far as I know once you go beyond a certain firmware version with polycom you cant go back. Dovid Dovid, what you are calling firmware Polycom refers to as the BootROM. At the begining of the release notes for whatever version of bootrom you want to use it will specify what versions you can upgrade from/downgrade to. In my experience what version of BootROM you are using is of little consequence since it only really affects methods of provisioning. I believe you can upgrade/downgrade to any version of the SIP application you want. As for what versions of the application work with asterisk, that depends on what features you are using. I'm running 1.6.7 on all our phones with no problems. However I'm not doing anything fancy with our system either. I've been using 2.1.0 on my home phone without any issues thus far, again not doing anything fancy. -Dave ___ --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] H323-to-SIP proxy
I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Firmware
Dovid B wrote: Doug is this for the sip version or firmware ? As far as I know once you go beyond a certain firmware version with polycom you cant go back. Sip 1.5.2 Bootrom 3.1.3 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
[asterisk-users] Net-talk
I wanted to try and see if I could get my Hawkings Net-Talk USB phone to work with our asterisk setup via yakaphone. Has anyone ever tried this? It sees the mic and speakers, but if we could get the keypad to talk with yaka and in turn with asterisk, that would be really nice. If there are any other recommendations for a VOIP USB phone that you could plug into a windows and/or linux machine and use it with either a iax program on the phone, or on the pc itself... Rob ___ --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] Billing Telephone Number (BTN)
Forrest Beck wrote: I have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. Yes, CallerID(num) should work. I had this issue when setting my outbound caller ID to a toll free number and trying to dial a few other toll free numbers. The call could not be completed because they had no way to know how to bill the call. Setting outbound callerID(num) to a regular toll number fixed it. Thanks, Steve Totaro ___ --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 - Streaming Audio Bridge
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ICES Lee Archer wrote: I used mpg123 to stream air traffic control as a MOH class but I also found it didn't always work with the shoutcast servers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Germann Sent: 27 February 2007 02:17 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Asterisk - Streaming Audio Bridge Greetings, Does anyone know of a tool that can act as a VoIP client and stream to a streaming server such as shoutcast/icecast, etc. I've got a client interested in doing basketball play by plays during tourney season. They have * in place now and the bandwidth to burn for streaming out. In the old world, I did an analog phone patch - mixer - encoder - streaming server. What I'm thinking of is more along the lines of a client that registers as a SIP/IAX client, answers the phone and patches it to a streaming server. Thoughts/suggestions? Thanks Eric ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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] H323-to-SIP proxy
T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --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] Long call setup times on SIP to zaptel calls
One thing I've noticed with SIP - ZAP calls for quite some time is that when asterisk is dialing n digits out the zap line, it dials n-1 digits, pauses for *TWO* seconds, and then sends the nth digit. Doesn't matter how many numbers I want to send out the ZAP channel, this always seems to happen. Asterisk 1.2.x is affected for sure. I haven't tested 1.4 yet. But if we could get this figured out, that would shave two seconds off MY nearly-five-second setup time. Mojo P.S. Polycom Soundpoint 501, TDM w/ 4xFXO, Asterisk 1.2.13 Jordan Novak wrote: I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak ___ --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] running asterisk through cellphone
hi dovid, thx for replying, as I can see the chan_cellphone patch was done by you, great! looks like this is exactly what I want. my goal is to connect a normal consumer cellphone to the asterisk-server, allowing anyone else to phone-in from their regular phone. it would be even better if I could use this setup to emulate extension - so lets assume my cellphone-number is 004369912345678, than I would like to have 3 separate extensions at 004369912345678-01, 004369912345678-02 and 004369912345678-03. is this possible? as I'm going to buy a separate phone for this task, can anyone recommend certain models (besides the RIM blackberry mentioned in the docs)? greetings, michael On 2/27/07, Dovid B [EMAIL PROTECTED] wrote: What is the cellular connection for ? Are you using this for inbound or the clients will call in in from thier cell phones ? If you need incoming (and or ourgoing) lines you can get one from an ITSP. If you want to use your cell phone you can use chan_cellphone. In order to use it you will need to install the patch. For more information have at look at this: http://bugs.digium.com/view.php?id=8919 http://bugs.digium.com/view.php?id=8919 - Original Message - *From:* Michael Kamleitner [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Tuesday, February 27, 2007 6:54 PM *Subject:* [asterisk-users] running asterisk through cellphone hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via data-cable (or bluetooth?) and use this as the single line to call-in. searching the asterisk-forums I found mentions of chan_cellphone, which is probably a patch for exactly this kind of usage, right? I'ld be thankful if you could just point me to the right direction (I'm quite new to asterisk). thx in advance! 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 ___ --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 -- 10 Jahre The Gap Party am 15.3.2007 - www.tengap.at Mag. Michael Kamleitner - [EMAIL PROTECTED] +43 699 11607923 https://www.xing.com/profile/Michael_Kamleitner - m-otion GmbH Favoritenstr 4-6/III, 1040 Wien +43 1 205705 / 21 (Fax 99) - www.m-otion.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323-to-SIP proxy
What about the SIP leg? - Mensaje Original - De: Michelle Dupuis [EMAIL PROTECTED] Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviados: martes 27 de febrero de 2007 16h'08 (GMT-0300) America/Argentina/Buenos_Aires Asunto: RE: [asterisk-users] H323-to-SIP proxy T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --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
Re: [asterisk-users] Long call setup times on SIP to zaptel calls
From: Mojo with Horan Company, LLC [EMAIL PROTECTED] Date: Tue, 27 Feb 2007 10:18:51 -0900 One thing I've noticed with SIP - ZAP calls for quite some time is that when asterisk is dialing n digits out the zap line, it dials n-1 digits, pauses for *TWO* seconds, and then sends the nth digit. Doesn't seem to happen in TDM400P and X100P cards, though. Could it be some feature configured in your particular card? Yuan Liu Doesn't matter how many numbers I want to send out the ZAP channel, this always seems to happen. Asterisk 1.2.x is affected for sure. I haven't tested 1.4 yet. But if we could get this figured out, that would shave two seconds off MY nearly-five-second setup time. Mojo P.S. Polycom Soundpoint 501, TDM w/ 4xFXO, Asterisk 1.2.13 Jordan Novak wrote: I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak ___ --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] Long call setup times on SIP to zaptel calls
On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote: I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak I'm not sure if it's related, but we are doing only SIP to SIP calling with Asterisk 1.4 and experience the same thing. The signaling shows up instantly, but it takes 5-7 seconds before ringback is heard. Watching the CLI it does look like it takes a long time for the channel to pick up an dial. regards, David ___ --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] Long call setup times on SIP to zaptel calls
Mojo with Horan Company, LLC wrote: One thing I've noticed with SIP - ZAP calls for quite some time is that when asterisk is dialing n digits out the zap line, it dials n-1 digits, pauses for *TWO* seconds, and then sends the nth digit. Doesn't matter how many numbers I want to send out the ZAP channel, this always seems to happen. Asterisk 1.2.x is affected for sure. I haven't tested 1.4 yet. But if we could get this figured out, that would shave two seconds off MY nearly-five-second setup time. This seems to happen with ALL dialing out non-PRI FXO ports and has happened since at least 1.2.0 and maybe long before that. Irritating as hell because when outpulsing 11 digits to the PSTN, every second counts as the user gets annoyed with 11 x 300ms of silence. Yes, the default for Asterisk is 100ms DTMF, but many systems don't recognize DTMFs that short. As far as I know it has not been fixed. ___ --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] Long call setup times on SIP to zaptel calls
Yuan LIU wrote: From: Mojo with Horan Company, LLC [EMAIL PROTECTED] Date: Tue, 27 Feb 2007 10:18:51 -0900 One thing I've noticed with SIP - ZAP calls for quite some time is that when asterisk is dialing n digits out the zap line, it dials n-1 digits, pauses for *TWO* seconds, and then sends the nth digit. Doesn't seem to happen in TDM400P and X100P cards, though. Could it be some feature configured in your particular card? Use ZapBarge to monitor your TDM400P or X100P FXO port before sending a call out that port. I think you'll hear the delay. ___ --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] Authentication Command
What version of Asterisk are you running? The 2 seconds of silence before it moves forward is probably because you haven't set digittimeout and or you are not hitting # when you finish entering your password. What does your dialplan look like that is calling the authenticate command. Please give more information. On 2/27/07, Supa [EMAIL PROTECTED] wrote: Anyone else experiencing a slow authentication command. I noticed this command takes about 1.5 - 2 seconds of silence before it asked for password, then another 2 sec of silence before it moves froward after that. Any ideas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] chan_sip.c:10173 handle_response: Dont know how tohandle a 202 Accepted respons
Looks like asterisk is receiving 202 while it is not expecting it. /*! \brief Handle SIP response in dialogue */ /* XXX only called by handle_request */ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno) Can you provide ethereal capture when you see this log message? Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Tuesday, February 27, 2007 1:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_sip.c:10173 handle_response: Dont know how tohandle a 202 Accepted respons What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk CDR and mysql
Hello all,
I added a record named pre_dst in the cdr table.
It has the same type as dst field.
And I used this line in the dialplan:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
When I call, 70123456, (7 is only to use the provider trunk),
I have this in the CLI:
Executing Set(SIP/foo-0816a490, CDR(predst)=0123456) in new stack
-- Executing NoOp(SIP/foo-0816a490, 0123456) in new stack
-- Executing Dial(SIP/moi-0816a490, SIP/[EMAIL PROTECTED]) in new stack
But nothing in the pre_dst field in cdr table.
Is there something wrong I did?
Thanks in advance.
___
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[asterisk-users] TE212P on FC6 - stack overflow?
Hi all did anyone of you experience an error like do_irq: stack overflow in configuring a TE212P on Fedora core 6? The server immediately hangs, I don't know if this can be related to hardware configuration or kernel incompatibility... This problem arises when I try to configure the channels with the usual command ztcfg and it is strictly related to the presence of the echo canceller onboard. Thanks a lot Marco Looking for earth-friendly autos? Browse Top Cars by Green Rating at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/___ --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] TE212P on FC6 - stack overflow?
Have you tried starting Linux with irqpoll / noapic? Sounds like a BIOS bug.. MD _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Parisotto Sent: Tuesday, February 27, 2007 3:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TE212P on FC6 - stack overflow? Hi all did anyone of you experience an error like do_irq: stack overflow in configuring a TE212P on Fedora core 6? The server immediately hangs, I don't know if this can be related to hardware configuration or kernel incompatibility... This problem arises when I try to configure the channels with the usual command ztcfg and it is strictly related to the presence of the echo canceller onboard. Thanks a lot Marco _ No need to miss a message. Get email http://us.rd.yahoo.com/evt=43910/*http://mobile.yahoo.com/mail on-the-go with Yahoo! Mail for Mobile. Get http://us.rd.yahoo.com/evt=43910/*http://mobile.yahoo.com/mail started. ___ --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] Authentication Command
I am using 1.2.3 I get a 2 second pause before get the auth command from my primary dial plan On 2/27/07, Matt [EMAIL PROTECTED] wrote: What version of Asterisk are you running? The 2 seconds of silence before it moves forward is probably because you haven't set digittimeout and or you are not hitting # when you finish entering your password. What does your dialplan look like that is calling the authenticate command. Please give more information. On 2/27/07, Supa [EMAIL PROTECTED] wrote: Anyone else experiencing a slow authentication command. I noticed this command takes about 1.5 - 2 seconds of silence before it asked for password, then another 2 sec of silence before it moves froward after that. Any ideas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] H323-to-SIP proxy
T.38 pass-through should work fine on the SIP leg. (With Asterisk 1.40) There are a few bugs but you can get past them. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 2:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] H323-to-SIP proxy What about the SIP leg? - Mensaje Original - De: Michelle Dupuis [EMAIL PROTECTED] Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviados: martes 27 de febrero de 2007 16h'08 (GMT-0300) America/Argentina/Buenos_Aires Asunto: RE: [asterisk-users] H323-to-SIP proxy T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --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
Re: [asterisk-users] Polycom Firmware
Sip 1.5.2 Bootrom 3.1.3 Anyone know any good reasons NOT to use the latest? I believe Bootrom 3.2.2 B and Firmware (SIP) 2.1.0. It's also the stuff the new IP 650's are using. -Kenneth ___ --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] SER / IAX solution
I find IAX connection with FWD very unreliable so I think I'll have to roll out my own SIP Express Router as I want to communicate with few SIP clients. So I hope this the right solution. I'm new to SER and to my understanding SER is like a road-map it tells the SIP Clients where they are so they can communicate directly with each other without going through a central server, am I right? What is the equivalent solution for IAX? If I have 5 clients registered to my box and all of them want to talk to each other the connection would go through my Asterisk server and that is not acceptable as they will kill my upload bandwidth; I want them to communicate with each other. What is FWD using for IAX clients? -- #Joseph ___ --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] SER / IAX solution
If re-invites are allowed then once both IAX endpoints are connected to Asterisk and the call is active the server will attempt to step out of the call. This actually works for both sip and IAX. On 2/27/07, Joseph [EMAIL PROTECTED] wrote: I find IAX connection with FWD very unreliable so I think I'll have to roll out my own SIP Express Router as I want to communicate with few SIP clients. So I hope this the right solution. I'm new to SER and to my understanding SER is like a road-map it tells the SIP Clients where they are so they can communicate directly with each other without going through a central server, am I right? What is the equivalent solution for IAX? If I have 5 clients registered to my box and all of them want to talk to each other the connection would go through my Asterisk server and that is not acceptable as they will kill my upload bandwidth; I want them to communicate with each other. What is FWD using for IAX clients? -- #Joseph ___ --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 -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Saving Dialplan in CLI
Is there anyway to unset the extensions.conf definition of writeprotect=yes while in the CLI interface (or by other mechanism) to enable the dialplan save command? I accidentally overwrote my extensions.conf but still have a running copy of asterisk with the old dial plan running in memory. while it would not be difficult for me to rebuild what I lost - it would be easier if I could just save it from the running copy. I will definitely set the writeprotect option to no in the future. Feedback Appreciated, -- --- John C. Wolosuk Jr. Unix/Linux Systems Administrator Academic Computing Communications Center University of Illinois @ Chicago E-Mail: jwolosuk at uic dot edu --- ___ --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] chan_sip.c:10173 handle_response: Dont know howtohandle a 202 Accepted respons
From: Bala Neelakantan [EMAIL PROTECTED]
Date: Tue, 27 Feb 2007 14:21:32 -0600
Looks like asterisk is receiving 202 while it is not expecting it.
/*! \brief Handle SIP response in dialogue */
/* XXX only called by handle_request */
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct
sip_request *req, int ignore, int seqno)
Can you provide ethereal capture when you see this log message?
Neel,
Thanks for the reply. I don't have ethereal on the machine and not sure how
to capture - non-graphic terminal environment. Below is output from
tcpdump. In this session, I see two 202 Accepted from 1.4.0, only one
don't know notice. Interestingly, identical tests between two 1.2.13
Asterisk does not produce this.
I assume that this is nothing serious, because the session completes without
any problem, and the message is only a notice. If anything, I'll simply
revert to 1.2. (These are non-production.)
Yuan Liu
13:42:12.685850 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 749
E.. [EMAIL PROTECTED]
..
...INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1
13:42:12.686783 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 430
E...D[EMAIL PROTECTED] +
...
..
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.10:5060;br
13:42:12.687705 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 710
E...D'[EMAIL PROTECTED]
...
..
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.10:5060;branch
13:42:12.688229 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 363
[EMAIL PROTECTED]
..
s.WACK sip:[EMAIL PROTECTED] SIP/2.0
V
ia: SIP/2.0/UDP 10.0
13:42:12.761105 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 371
E...D([EMAIL PROTECTED] d
...
..
.{N/MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/U
13:42:12.761685 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 468
E:[EMAIL PROTECTED]
..
...SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.0.0.201:5060
13:42:12.793347 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 399
E;[EMAIL PROTECTED]
..
..{MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
V
ia: SIP/2.0/UDP
13:42:12.793863 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 448
E...D)[EMAIL PROTECTED] .
...
..
...oSIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.0.0.10:5060;
13:42:12.796133 IP 10.0.0.201.5060 10.0.0.10.5060: UDP, length: 332
[EMAIL PROTECTED] .
...
..
.Tw.BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1
13:42:12.796777 IP 10.0.0.10.5060 10.0.0.201.5060: UDP, length: 463
E[EMAIL PROTECTED]
..
...SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.201:5060;branc
Thanks,
Neel
-Original Message-
What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from
1.4.0.)
Yuan Liu
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Re: [asterisk-users] running asterisk through cellphone
On Tue, 27 Feb 2007, Michael Kamleitner wrote: hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via data-cable (or bluetooth?) and use this as the single line to call-in. searching the asterisk-forums I found mentions of chan_cellphone, which is probably a patch for exactly this kind of usage, right? I'ld be thankful if you could just point me to the right direction (I'm quite new to asterisk). thx in advance! If I understand you, you want to call the mobile phone, and have asterisk deal with the audio? the only think I know of is Dock'n'Talk http://www.phonelabs.com/prd05.asp but that has an analogue output, so you'd need analogue into asterisk, and if you have analogue in, then you might as well use a landline if you can... Gordon ___ --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] Net-talk
On Tue, 27 Feb 2007, Rob Schall wrote: I wanted to try and see if I could get my Hawkings Net-Talk USB phone to work with our asterisk setup via yakaphone. Has anyone ever tried this? It sees the mic and speakers, but if we could get the keypad to talk with yaka and in turn with asterisk, that would be really nice. If there are any other recommendations for a VOIP USB phone that you could plug into a windows and/or linux machine and use it with either a iax program on the phone, or on the pc itself... I used a Yealink USB phone device with reasonable success... (and there's linux drivers for it) Gordon ___ --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] Saving Dialplan in CLI
John C. Wolosuk Jr. wrote: Is there anyway to unset the extensions.conf definition of writeprotect=yes while in the CLI interface (or by other mechanism) to enable the dialplan save command? I accidentally overwrote my extensions.conf but still have a running copy of asterisk with the old dial plan running in memory. show dialplan might be your friend but the output is not an executable dialplan. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE212P on FC6 - stack overflow?
Hi Michelle, actually, I didn't try it... The server is a HP Proliant ML150T G3. Currently I'm not in the condition to follow your suggestion, but I hope in the near future to be able to give you a feedback. Thanks! Marco Have you tried starting Linux with irqpoll / noapic? Sounds like a BIOS bug.. MD _ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Marco Parisotto Sent: Tuesday, February 27, 2007 3:27 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] TE212P on FC6 - stack overflow? Hi all did anyone of you experience an error like do_irq: stack overflow in configuring a TE212P on Fedora core 6? The server immediately hangs, I don't know if this can be related to hardware configuration or kernel incompatibility... This problem arises when I try to configure the channels with the usual command ztcfg and it is strictly related to the presence of the echo canceller onboard. Thanks a lot Marco Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta. Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097___ --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] Yellow or Red alarm on TE110P ????
hi after a many manipulation i get OK/YELLOW signal what does mean? ___ --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] Quintum configuration ASM200 Analog 2 tenor port
Hi, just wondering if there is anyone that can help me configure my quintum box to operate with asterisk. i have tried and made numerous attemtps configuring the tenor to work with [EMAIL PROTECTED] but have been unlucky. anyone out there has a cheat sheet to configure this device. thanks.. for some reason i cannot get it to work. your help is appreciated. _ Dont miss your chance to WIN 10 hours of private jet travel from Microsoft® Office Live http://clk.atdmt.com/MRT/go/mcrssaub0540002499mrt/direct/01/ ___ --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] Quintum configuration ASM200 Analog 2 tenor port
I just struggled through the config on a Tenor AX. I'm not sure I can help but I'll try. What do you need to do? -Steve FRANCISCO PEREZ-LANDAETA wrote: Hi, just wondering if there is anyone that can help me configure my quintum box to operate with asterisk. i have tried and made numerous attemtps configuring the tenor to work with [EMAIL PROTECTED] but have been unlucky. anyone out there has a cheat sheet to configure this device. thanks.. for some reason i cannot get it to work. your help is appreciated. _ Don’t miss your chance to WIN 10 hours of private jet travel from Microsoft® Office Live http://clk.atdmt.com/MRT/go/mcrssaub0540002499mrt/direct/01/ ___ --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] Yellow or Red alarm on TE110P ????
younss azzayani wrote: hi after a many manipulation i get OK/YELLOW signal what does mean? Don't manipulate. :-P Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] jittery audio in voiceprompts
Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple hello world dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under vmware and both exhibited this issue. Linux box is perhaps a little under powered, it is an Intel Celeron 467Mhz. I tested with asterisk niced to -18, which did not change the problem. I am using a Linksys SPA942 sip phone, using ALAW. I have tried gsm and alaw prompts but that didn't solve it. I am running Linux blue 2.6.16.20 on a debian stable machine. I have been compiling and installing asterisk from source. I have tried looking at the debug messages in asterisk but nothing seems to indicate an issue. I read somewhere that disabling X can help, but it did not in my case. I am at a loss as to how I might track down the problem and fix it. Any pointers would be greatly appreciated. Thanks, Jason ___ --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: Authentication Command
Anyone else experiencing a slow authentication command. I noticed this command takes about 1.5 - 2 seconds of silence before it asked for password, then another 2 sec of silence before it moves froward after that. Any ideas I use Authentication regularly, no delay at all. Post your dialplan example, * version, hardware, more info about your setup. Make sure you answer the call first, before you invoke the authenticate cmd. JR -- JR Richardson Engineering for the Masses ___ --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: Polycom Firmware
Anyone know any good reasons NOT to use the latest? I believe Bootrom 3.2.2 B and Firmware (SIP) 2.1.0. It's also the stuff the new IP 650's are using. I'm using 1.6.6 with no issues, besides the known call transfer thing. I tried 2.X on a IP_601 and had trouble with the buddy-watch presence, had to shelf the phone and go with a 601 with 1.6.6. That's the only thing I'm aware of is presence seemed to break with the latest firmware. JR -- JR Richardson Engineering for the Masses ___ --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] jittery audio in voiceprompts
On Wed, 2007-02-28 at 10:12 +1100, Jason Lewis wrote: Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple hello world dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under vmware and both exhibited this issue. Linux box is perhaps a little under powered, it is an Intel Celeron 467Mhz. I tested with asterisk niced to -18, which did not change the problem. I am using a Linksys SPA942 sip phone, using ALAW. I have tried gsm and alaw prompts but that didn't solve it. I am running Linux blue 2.6.16.20 on a debian stable machine. I have been compiling and installing asterisk from source. I have tried looking at the debug messages in asterisk but nothing seems to indicate an issue. I read somewhere that disabling X can help, but it did not in my case. I am at a loss as to how I might track down the problem and fix it. Any pointers would be greatly appreciated. Thanks, Jason-- What do you have installed, that will provide the 1Khz timing interrupts you will need to function properly? murf smime.p7s Description: S/MIME cryptographic signature ___ --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] jittery audio in voiceprompts
Steve Murphy wrote: On Wed, 2007-02-28 at 10:12 +1100, Jason Lewis wrote: I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple hello world dial plan. What do you have installed, that will provide the 1Khz timing interrupts you will need to function properly? Err.. I was not aware I would have to install anything to do that. I guess that could mean I have nothing installed. What should I have installed? Thanks, Jason ___ --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] Yellow or Red alarm on TE110P ????
younss azzayani wrote on February 27, 2007 2:30 AM the cable is a simple cable break or: the cable schema we see bellow 1. If a piece of equipment such as the TE110P card is NOT seeing a T1 signal coming in, it will go into red alarm. That same piece of equipment will then output on it's transmit pins a yellow alarm signal. 2. If a piece of equipment sees a yellow alarm signal coming in, that piece of equipment will put itself into yellow alarm. These alarms are very useful for trouble shooting, especially if long cables (1000s of feet) or several connections are involved. So, take care of the red alarm first (verify that a valid signal is coming in) and the yellow alarm will no longer be sent out and you won't be getting the yellow alarm message. (1-4 2-5): That is correct for most equipment. However there are a few pieces of equipment that need a straight through cable (1-1, 2-2, 4-4 and 5-5). For example, our local telco's 'network interface unit' to our Digium T1 cards uses a straight through cable. I hope this helps. Don Pobanz ___ --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] Yellow or Red alarm on TE110P ????
Philipp Kempgen wrote: younss azzayani wrote: hi after a many manipulation i get OK/YELLOW signal what does mean? Don't manipulate. :-P Regards, Philipp Start Asterisk and turn on the proper debugging. Thanks, Steve ___ --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] Saving Dialplan in CLI
Philipp Kempgen wrote: John C. Wolosuk Jr. wrote: Is there anyway to unset the extensions.conf definition of writeprotect=yes while in the CLI interface (or by other mechanism) to enable the dialplan save command? I accidentally overwrote my extensions.conf but still have a running copy of asterisk with the old dial plan running in memory. show dialplan might be your friend but the output is not an executable dialplan. Regards, Philipp A Ciscoesque show command, show running-configuration would be pretty cool. ___ --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] jittery audio in voiceprompts
Isn't there a zap dummy (or something that uses the RTC) included in Asterisk 1.40 that creates the timing source? We don't install any external timing sources and we don't have choppyness problems on pure sip connections... Jason - is this on a standard PC motherboard (or a mini device like Linksys WRT)? MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Tuesday, February 27, 2007 6:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] jittery audio in voiceprompts On Wed, 2007-02-28 at 10:12 +1100, Jason Lewis wrote: Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple hello world dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under vmware and both exhibited this issue. Linux box is perhaps a little under powered, it is an Intel Celeron 467Mhz. I tested with asterisk niced to -18, which did not change the problem. I am using a Linksys SPA942 sip phone, using ALAW. I have tried gsm and alaw prompts but that didn't solve it. I am running Linux blue 2.6.16.20 on a debian stable machine. I have been compiling and installing asterisk from source. I have tried looking at the debug messages in asterisk but nothing seems to indicate an issue. I read somewhere that disabling X can help, but it did not in my case. I am at a loss as to how I might track down the problem and fix it. Any pointers would be greatly appreciated. Thanks, Jason-- What do you have installed, that will provide the 1Khz timing interrupts you will need to function properly? murf ___ --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] TE110P: Error == Asterisk died with code 1.
Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf [channels] context=from-pstn switchtype=national signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=yes group=0 signalling=pri_cpe context = from-pstn channel =1-23 == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not registering Port with VSP
Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been told that my asterisk server is registering my IP with the VSP but the port is empty. ___ --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] Voice mail is not giving unavailable or busy prompts
Hi: This should be easy. I'm running 1.2.15. When a caller calls someone's voice mail, it goes straight to a beep, even though there is an unavail.wav file in that user's voice mail directory. Here is the relevant part of extensions.conf: [internal] exten = 2211,1,Dial(SIP/211,10) exten = 2211,2,VoiceMail([EMAIL PROTECTED]) exten = 2211,3,Hangup Here is the relevant part of voicemail.conf: [default] 211 = ,Mr Test,[EMAIL PROTECTED] Here's what I see in the console: -- Executing Dial(SIP/210-081990b0, SIP/211|10) in new stack -- Called 211 -- SIP/211-0819e5f0 is ringing -- Nobody picked up in 1 ms -- Executing VoiceMail(SIP/210-081990b0, [EMAIL PROTECTED]) in new stack -- Playing '/var/spool/asterisk/voicemail/default/211/unavail' (language 'en') -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/211/tmp/EWtUPC format: wav, 0x81a3c98 -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Executing Hangup(SIP/210-081990b0, ) in new stack == Spawn extension (internal, 2211, 3) exited non-zero on 'SIP/210-081990b0' This is what is actually in /var/spool/asterisk/voicemail/default/211: asterisk1 211 # ls -liah total 108K 4918844 drwx-- 7 root root 4.0K Feb 27 17:59 . 4898961 drwxr-xr-x 5 root root 4.0K Feb 27 17:05 .. 4918846 drwx-- 2 root root 4.0K Feb 27 18:32 INBOX 4918850 drwx-- 2 root root 4.0K Feb 27 17:12 Old 4918849 -rwx-- 1 root root 56K Feb 27 17:10 busy.wav 4918845 drwx-- 2 root root 4.0K Feb 27 17:05 temp 4918847 drwx-- 2 root root 4.0K Feb 27 18:32 tmp 4931585 drwxr-xr-x 2 root root 4.0K Feb 27 17:59 unavail 4918848 -rwx-- 1 root root 20K Feb 27 17:13 unavail.wav Asterisk creates that unavail directory after the first time someone tries to call in. Ideas? -Stephen- ___ --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] TE110P: Error == Asterisk died with code 1.
Jeronimo Romero wrote: Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf [channels] context=from-pstn switchtype=national signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=yes group=0 signalling=pri_cpe context = from-pstn channel =1-23 == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == tail /var/log/asterisk/full will give you good insight. Thanks, Steve ___ --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] jittery audio in voiceprompts
Michelle Dupuis wrote: Isn't there a zap dummy (or something that uses the RTC) included in Asterisk 1.40 that creates the timing source? We don't install any external timing sources and we don't have choppyness problems on pure sip connections... Yes, I have been looking into that after reading Steve's response. Unfortunately I get a compile error with it. I'll try a newer kernel. I have a pure SIP installation also Jason - is this on a standard PC motherboard (or a mini device like Linksys WRT)? Yes, standard PC (although older as mentioned in previous post) Thanks, Jason ___ --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
