Re: [asterisk-users] Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)
Thanks, I already found these names, but maybe I missed some ! Thanks again, JM On 5/14/07, Yossi Ben Hagai [EMAIL PROTECTED] wrote: Check rtpproxy from portone for media proxy and nat traversal. http://www.voip-info.org/wiki/view/Portaone+rtpproxy another option is the MediaProxy from AG projects: http://www.voip-info.org/wiki-MediaProxy Joss. On 5/11/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://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] Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)
Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM ___ --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] SIPGetHeader in Asterisk 1.4
Hi to all, I recently tried to upgrade my Asterisk 1.2 to 1.4. I use quite extensively SIPGetHeader cmd in my Dialplan. But this application is not found in 1.4.2, and I do not see it in 1.4.4code either ??? I could find indeed SIPAddHeader in code. BUT Where did SIPGetHeader go ? any new cmd replacing this one ? Thanks, Jean-Marc ___ --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] SIPGetHeader in Asterisk 1.4
Thanks ! On 5/2/07, Manu Mehta [EMAIL PROTECTED] wrote: Hi, You can use function SIP_HEADER instead. See http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header *Manu Mehta* * * *A R I C E N T* Plot-17, Sector 18, Gurgaon 122015, Haryana, India Main +91.124.4095888 x3274 Fax +91.124.4095912 *Jean-Marc Salsa [EMAIL PROTECTED]* Sent by: [EMAIL PROTECTED] 05/02/2007 07:03 PM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com cc Subject [asterisk-users] SIPGetHeader in Asterisk 1.4 Hi to all, I recently tried to upgrade my Asterisk 1.2 to 1.4. I use quite extensively SIPGetHeader cmd in my Dialplan. But this application is not found in 1.4.2, and I do not see it in 1.4.4code either ??? I could find indeed SIPAddHeader in code. BUT Where did SIPGetHeader go ? any new cmd replacing this one ? Thanks, Jean-Marc___ --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 *** Aricent-Unclassified *** DISCLAIMER: This message is proprietary to Aricent and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are strictly prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus. ___ --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] Returning different SIP Hangup Cause
Hi, I would like to return different values/cause to another SIP Server with Hangup cmd. I tried to put different values in Hangup(xx) ... but it always returns the same value ! How can I send back different error cause ? Thanks, Jean-Marc ___ --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] Voicemail: How to send a notification even if Caller does not let any messages?
Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc ___ --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] Voicemail: How to send a notification even if Caller does not let any messages?
Thanks for the answer, I already use this extapp, to set on another server the MWI. But how to know if user has not let a message ? One could guess that 0 is the number of message to trigger such a notification ... but 0 is the number of message as well when you erase all your messages, so you shouldn't send a notification in that case. Any idea please ? Thanks, JM On 4/16/07, Rob Schall [EMAIL PROTECTED] wrote: If the voicemail portion is reached, but hungup on, the extapp portion of the config file is still executed. So you could have an external app which does any number of things (IM, etc). Rob Jean-Marc Salsa wrote: Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages ? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc -- ___ --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] Voicemail: How to send a notification even if Caller does not let any messages?
And by the way, I forgot, If I remember carefully, there is not so much info passed to this script (VM Number, context Number of messages) ... So for example, how do you get the caller ID info ? Thanks again, JM On 4/16/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Thanks for the answer, I already use this extapp, to set on another server the MWI. But how to know if user has not let a message ? One could guess that 0 is the number of message to trigger such a notification ... but 0 is the number of message as well when you erase all your messages, so you shouldn't send a notification in that case. Any idea please ? Thanks, JM On 4/16/07, Rob Schall [EMAIL PROTECTED] wrote: If the voicemail portion is reached, but hungup on, the extapp portion of the config file is still executed. So you could have an external app which does any number of things (IM, etc). Rob Jean-Marc Salsa wrote: Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages ? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail: How to send a notification if Caller hags up during announcement
Hi, I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc ___ --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] Issue with Calling Name ID in SIP: Asterisk sets Caller ID Number as Name if NO Name
Hi, fyi, I use Asterisk 1.2.9.1 In some scenarios, we receive call from PSTN without Callerd ID Name (which is normal). I would like to transfer this call to another softswitch. Again, I would like to let this this CallerID Name Empty. If I look at the logs, I can see -- Executing Macro(SIP/localdomain.com-b79242f0, set-callerid-name) in new stack -- Executing Set(SIP/localdomain.com-b79242f0, CALLERNAME_TMP=) in new stack -- Executing GotoIf(SIP/localdomain.com-b79242f0, 1?format_empty|1) in new stack -- Goto (macro-set-callerid-name,format_empty,1) -- Executing Set(SIP/localdomain.com-b79242f0, CALLERNAME=) in new stack -- Executing Goto(SIP/localdomain.com-b79242f0, set_callername|1) in new stack -- Goto (macro-set-callerid-name,set_callername,1) -- Executing Set(SIP/localdomain.com-b79242f0, CALLERID(name)=) in new stack So, it should be alright! Then I forward the call: -- Executing Dial(SIP/localdomain.com-b79242f0, SIP/990003726831598@next-hop|30) in new stack And if I look into SIP debug mode: Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 11 lines Reliably Transmitting (no NAT) to XXX:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDPXXX:5060;branch=z9hG4bK6dfcce4c;rport From: *0037253415630* sip:[EMAIL PROTECTED];tag=as4479803d To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: GSMDuo-VM Max-Forwards: 70 Date: Thu, 01 Mar 2007 11:05:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY P-Asserted-Identity: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 236 As you can see, name is set ! eventhough callerid(name)= Is it a bug ? How can I really clear this callerid(name) ? How to prevent Asterisk to put back as Name, the number ? Thanks for your kind return ! Regards, Jean -Marc -- Called [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] MixMonitor RingBack Tone Issue
Indeed, perfect ! Thanks a lot ... JM On 2/17/07, Trevor Peirce [EMAIL PROTECTED] wrote: Jean-Marc Salsa wrote: exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r mailto:SIP/[EMAIL PROTECTED],30,r) Everything works perfectly, except when the softswitch, or the PSTN sends back RingBack Tone. I can see the RTP flow arriving to Asterisk, but, it seems that Asterisk doesn't forward it to the other party (next-hop). Yes because you have the r in there, asterisk sends its own ringing. If you want ringing to be heard from the PSTN, you need to leave that option disabled. ___ --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] MixMonitor RingBack Tone Issue
Hi,
I use in Production : Asterisk 1.2.9.1
We Use Asterisk as a SIP Transit Server to record centrally all the calls.
The call flow would be:
incoming calls : PSTN - GW -SIP- Asterisk(Record) -SIP- Softswitch - IP
Phone
outgoing calls : IP Phone - Softswitch -SIP- Asterisk(Record) -SIP- GW
- PSTN
Dial plan in Asterisk is quite simple:
[record]
exten = s,1,Set(CALLFILENAME=${TIMESTAMP}-${UNIQUEID})
exten = s,n,Set(CALLERID(name)=${CALLERID(name)})
exten = s,n,Set(CALLERID(number)=00${CALLERID(number)})
exten = s,n,MixMonitor(${CALLFILENAME}.WAV,b)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r)
Everything works perfectly, except when the softswitch, or the PSTN sends
back RingBack Tone.
I can see the RTP flow arriving to Asterisk,
but, it seems that Asterisk doesn't forward it to the other party
(next-hop).
Any ideas why ?
How can I bypass this issue ?
Thanks,
Jean-Marc
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[asterisk-users] MWI across multiple servers
Jon, I would be as well very interested in your Voicemail Solution : AGI + Web Interface to retrieve voice messages. By the way, you sotre to MySQL, do you use ODBC for that ? or something else, in that case, what ;o) ? Thanks in advance ! Jean-Marc On 12/7/06, Jon Farmer [EMAIL PROTECTED] wrote: I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Porier, Jeremy M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 6 December, 2006 4:20:04 PM Subject: [asterisk-users] MWI across multiple servers We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --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 Send instant messages to your online friends http://uk.messenger.yahoo.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
[asterisk-users] cmd Record doesn't resume Dialplan if phone Hangs-Up.
Hi,
I have tried to use the Record Command in Asterisk,
Here is the configuration :
exten = record,1,Answer
...
exten = record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV)
exten = record,n,Playback(vm-goodbye)
exten = record,n,system(/usr/local/bin/send-recording.pl --to ${EMAILADDR}
--file /var/spool/asterisk/record/${CALLFILENAME}.WAV)
exten = record,n,Hangup
If I hung up the phone during recording,
then Message is well there, but Asterisk does not continue its way to the
system command to send me the file.
If I change the record command to detect a 2 sec silence.
Then, Asterisk hangs up correctly and DO send the file ...
Has anyone noticed something similar ?
Did I do something wrong ?
Can somebody help me ?
Thanks,
Jean-Marc
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[asterisk-users] Auth Issue using Asterisk as Voicemail AND as Normal SIP Extension.
Hi, My case is a little bit complicated. I would like to use my Asterisk Box for 2 different services/providers : - Voicemail server for one - SIP Registrar and Proxy for some other extensions The problem is that Voicemail service is for another provider which has defined Extension like ABC ... We are connected to them through a SIP Trunk. Everything works fine Except IF ABC is also defined in the sip.conf as one of the other Extensions of the second virtual provider ... Thus, Asterisk doesn't ask anymore password for the SIP Trunk, but for the SIP Extension . Which is WHAT I DO NOT WANT ! because as you might guess, the call to Voicemail will be rejected ! I have been thinking of implementing SIP Multi-Domain, Would it be the way ? Very hard to find good doc on the subject ... could someone point it out please ? How should I implement it ? what domain should I put in Asterisk General SIP config ? Should I use for each of the defined extension a setting specifying from which domain they belong ? Thanks a lot for your help ! JM ___ --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 Voicemail with ODBC Realtime Access
Sorry to re-post, but as noone has answeredme ...
Maybe somebody will this time :o)
Thanks !
JM
On 10/29/06, Jean-Marc Salsa wrote:
Hi
I was trying to have realtime voicemail working with ODBC Driver.Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page )
But I do not manage to make it work with ODBC.Outside Asterisk, ODBC works fine, I can access my databases tables !
Asterisk fails to start if I use pre-connect = yes in res_odbc.conf ( See errors below )If I do not use it, then, I get another error message : res_config_odbc.c: SQL Alloc Handle failed! when I try to access the voicemail.
Any ideas would be more than welcome !
Thanks !
Here is my config
/etc/odbcinst.ini[MySQL]Description = ODBC for MySQLDriver = /usr/lib/libmyodbc.soSetup = /usr/lib/libodbcmyS.soFileUsage = 1
/etc/odbc.ini[MySQLast]Description = MySQL ODBC Driver TestingDriver = MySQL#Socket = /var/run/mysqld/mysqld.sockTrace = YesTraceFile = odbc_mysql.logServer = localhost
User = asteriskuserPassword = amp109Database = asteriskrealtime#Option = 3Port = 3306
isql -v MySQLast and then help shows me my Tables correctly,SELECT queries work perfectly
/etc/asterisk/res_odbc.conf[asterisk]enabled = yesdsn = MySQLastusername = asteriskuserpassword = amp109;pre-connect = yes
/etc/asterisk/extconfig.conf[settings]voicemail = odbc,asterisk,voicemail_users
Modules in Asterisk :asterisk*CLI show modules like odbcModule Description Use Countres_config_odbc.so ODBC Configuration 1
res_odbc.so ODBC Resource 0cdr_odbc.so ODBC CDR Backend 03 modules loaded
Error Messages:
if pre-connect = yes is used in res_odbc.confOct 29 16:23:21 VERBOSE[15016] logger.c: MySQL RealTime driver loaded.Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_adsi.so]Oct 29 16:23:21 VERBOSE[15016]
logger.c: [res_adsi.so] = (ADSI Resource)Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_odbc.so]Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_odbc.so] = (ODBC Resource)Oct 29 16:23:21 VERBOSE[15016] logger.c
: == Parsing '/etc/asterisk/res_odbc.conf': Oct 29 16:23:21 VERBOSE[15016] logger.c: == Parsing '/etc/asterisk/res_odbc.conf': FoundOct 29 16:23:21 NOTICE[15016] res_odbc.c: registered database handle 'asterisk' dsn-[MySQLast]
Oct 29 16:23:21 NOTICE[15016] res_odbc.c: Connecting asterisk
And this is what I get on the standard error output:*** glibc detected *** malloc(): memory corruption: 0x09f4daf0 ***/usr/sbin/safe_asterisk: line 50: 14920 Aborted (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 134Asterisk exited on signal 6.Automatically restarting Asterisk.*** glibc detected *** malloc(): memory corruption: 0x0a023af0 ***/usr/sbin/safe_asterisk: line 50: 15016 Aborted (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 134Asterisk exited on signal 6.Automatically restarting Asterisk.
If no pre-connect, then I can start asterisk, but I get this error when I try to access VM:Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing VoiceMail(IAX2/telegrupp-4,
[EMAIL PROTECTED]|) in new stackOct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'
Oct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4' Oct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'Oct 29 16:24:43 WARNING[15262] res_config_odbc.c: SQL Alloc Handle failed!
Oct 29 16:24:43 WARNING[15262] app_voicemail.c: No entry in voicemail config file for '200' Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing Goto(IAX2/telegrupp-4, exit-FAILED|1) in new stack
Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Goto (macro-vm,exit-FAILED,1)Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing Playback(IAX2/telegrupp-4, im-sorryan-error-has-occured) in
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[asterisk-users] Asterisk Voicemail with ODBC Realtime Access
Hi
I was trying to have realtime voicemail working with ODBC Driver.Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page )
But I do not manage to make it work with ODBC.Outside Asterisk, ODBC works fine, I can access my databases tables !
Asterisk fails to start if I use pre-connect = yes in res_odbc.conf ( See errors below )If I do not use it, then, I get another error message : res_config_odbc.c: SQL Alloc Handle failed! when I try to access the voicemail.
Any ideas would be more than welcome !
Thanks !
Here is my config
/etc/odbcinst.ini[MySQL]Description = ODBC for MySQLDriver = /usr/lib/libmyodbc.soSetup = /usr/lib/libodbcmyS.soFileUsage = 1
/etc/odbc.ini[MySQLast]Description = MySQL ODBC Driver TestingDriver = MySQL#Socket = /var/run/mysqld/mysqld.sockTrace = YesTraceFile = odbc_mysql.logServer = localhost
User = asteriskuserPassword = amp109Database = asteriskrealtime#Option = 3Port = 3306
isql -v MySQLast and then help shows me my Tables correctly,SELECT queries work perfectly
/etc/asterisk/res_odbc.conf[asterisk]enabled = yesdsn = MySQLastusername = asteriskuserpassword = amp109;pre-connect = yes
/etc/asterisk/extconfig.conf[settings]voicemail = odbc,asterisk,voicemail_users
Modules in Asterisk :asterisk*CLI show modules like odbcModule Description Use Countres_config_odbc.so ODBC Configuration 1
res_odbc.so ODBC Resource 0cdr_odbc.so ODBC CDR Backend 03 modules loaded
Error Messages:
if pre-connect = yes is used in res_odbc.confOct 29 16:23:21 VERBOSE[15016] logger.c: MySQL RealTime driver loaded.Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_adsi.so]Oct 29 16:23:21 VERBOSE[15016]
logger.c: [res_adsi.so] = (ADSI Resource)Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_odbc.so]Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_odbc.so] = (ODBC Resource)Oct 29 16:23:21 VERBOSE[15016] logger.c
: == Parsing '/etc/asterisk/res_odbc.conf': Oct 29 16:23:21 VERBOSE[15016] logger.c: == Parsing '/etc/asterisk/res_odbc.conf': FoundOct 29 16:23:21 NOTICE[15016] res_odbc.c: registered database handle 'asterisk' dsn-[MySQLast]
Oct 29 16:23:21 NOTICE[15016] res_odbc.c: Connecting asterisk
And this is what I get on the standard error output:*** glibc detected *** malloc(): memory corruption: 0x09f4daf0 ***/usr/sbin/safe_asterisk: line 50: 14920 Aborted (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 134Asterisk exited on signal 6.Automatically restarting Asterisk.*** glibc detected *** malloc(): memory corruption: 0x0a023af0 ***/usr/sbin/safe_asterisk: line 50: 15016 Aborted (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 134Asterisk exited on signal 6.Automatically restarting Asterisk.
If no pre-connect, then I can start asterisk, but I get this error when I try to access VM:Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing VoiceMail(IAX2/telegrupp-4,
[EMAIL PROTECTED]|) in new stackOct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'Oct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'
Oct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'Oct 29 16:24:43 WARNING[15262] res_config_odbc.c: SQL Alloc Handle failed!Oct 29 16:24:43 WARNING[15262] app_voicemail.c: No entry in voicemail config file for '200'
Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing Goto(IAX2/telegrupp-4, exit-FAILED|1) in new stackOct 29 16:24:43 VERBOSE[15262] logger.c: -- Goto (macro-vm,exit-FAILED,1)
Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing Playback(IAX2/telegrupp-4, im-sorryan-error-has-occured) in
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[asterisk-users] Accessing MySQL DB to set variables in Asterisk
Hi, I have set up a fax to email capability on our Asterisk Server (which is used for voicemail, and fax2email only), and would like to improve it ! Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is defined. If it is, then it will set the Email address where to send the fax. And actually, it would be the same for voicemail, Our users have different extensions. And we want them to have only one Mailbox. Thus we can configure the primary extension as the main voicemail number, but we need to have all the secondaries extension to be sent to the primary one. Thus I would like to seach into a DB, ifsuch a secondaryextension exists, it sends back the primary extension, so we can route the call to the appropriate mailbox ( primary ) I would like to use Asterisk + MySQL Realtime. I have set up in res_mysql.conf the way to access MySQL I have set up in extconfig.conf : fax2email = mysql,asteriskrealtime,fax2email And hen in my DB : DB Name = asteriskrealtime Table Name = fax2email ___ --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] Accessing MySQL DB to set variables in Asterisk
Hi, ( Sorry for previous post, it was incomplete :o( I have set up a fax to email capability on our Asterisk Server (which is used for voicemail, and fax2email only), and would like to improve it ! Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is defined. If it is, then it will set the Email address where to send the fax. I do not want to use extension.conf ... And actually, it would be the same for voicemail, Our users have different extensions. And we want them to have only one Mailbox. Thus we can configure the primary extension as the main voicemail number, but we need to have all the secondaries extension to be sent to the primary one. Thus I would like to seach into a DB, ifsuch a secondaryextension exists, it sends back the primary extension, so we can route the call to the appropriate mailbox ( primary ) I would like to use Asterisk + MySQL Realtime. I have set up in res_mysql.conf the way to access MySQL I have set up in extconfig.conf : fax2email = mysql,asteriskrealtime,fax2email And then in my DB : DB Name = asteriskrealtime Table Name = fax2email Table fax2email: Field EXT which contains extension numbers Field email which contains the email where to send the fax If I use DBGet, how to specify that I want to retrieve the email address from fax2email table, which matches the extension in Asterisk ? Thanks for your help guys ! Yours, Jean-Marc ( can cannot send more than 10 lines in one email :o) ___ --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] Accessing MySQL DB to set variables in Asterisk
Thanks, it seems to be not easy to use, but ... should do what's needed ! Thanks. On 10/16/06, Andrea Spadaccini [EMAIL PROTECTED] wrote: Ciao Jean-Marc, Everytime Asterisk receives a fax, I would like it to go and search in a DB if the Extension is defined. If it is, then it will set the Email address where to send the fax.You can use app_addon_mysql for your purposes.See: http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL .HTH,--Andrea SpadacciniMultimedia Technologies Institute s.r.l.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hints
I am using IP10s also It was working fine, but you needed to go into telnet mode, to activate the busy lamp, with the hint option ... moreover, if you wanted to pick up the phone call, then you needed also to add another telnet command to handle this pickup ! I know that swissvoice has now build 20, which allows all this through the web GUI interface ! Hope, this helps ! JM On 2/24/06, Alex Barnes [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Garth van Sittert Sent: 24 February 2006 07:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk hints Hi Mike I have build 18 on the IP10's and I have tried call-limit=1 but it still doesn't work. Do the extension phones need to have any settings changed to enablethis feature?I could be wrong but I think setting call-limit breaks hints in 1.2.x This is what finally forced me to get to grips with the GROUP() commandsfor limiting calls.Can't help much more than that though as we use Snom's with hints.HTHAlexInformation contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation.All unauthorized use, disclosure or distribution is strictly prohibited.If you are not the addressee, please notify the sender immediately and destroy all copies of this email.Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding.Thank you. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] DTMF Mode supported by VoiceMail Application
Hi, I would like to use Asterisk as VoiceMail system ... the only issue I have is with DTMF recognition. Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ... inband : works, but has some glitch ... not always good ... don't know why. rfc2833 : doesn't seem to work .. info : said to be not working ( cf http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode) Did anyone succeed that ? Thanks a lot ! JMS ___ --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] DTMF Mode supported by VoiceMail Application
Thanks, But, I do not have phones connected to Asterisk ... but only one peer : my softswitch ... So call flow is Phone - Softswitch - Asterisk - Voicemail Ican force the link Sofswitch - Asterisk ( Codec and DMTF Mode ) Codec is PCMx ... but as i said inband config is not working all the time ! Let me know if you think something else ... JMS On 2/22/06, Fabian Müller [EMAIL PROTECTED] wrote: Jean-Marc Salsa [EMAIL PROTECTED] writes: Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ...This depends on what your users are using. If you are using a Grandstream device you can configure in its administration interfacewhich dtmf mode the telefone should use. If your IP phone isconfigured to use rfc2833 for example then you would writedtmfmode=rfc2833 in your sip.conf. If all users use the samedtmfmode it should be ok to write this to the general section.Fabian Müller___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] ARI 0.06
You are wonderful !!! for this bug, I noticed later on that by removing the second path in the monitor folder ... I didn't get any error ... the script was searching inside a file, thinking that it could be a directory where recordings were. Anyway, Again, Thanks a lot, JM On 2/17/06, Dan Littlejohn [EMAIL PROTECTED] wrote: On 2/17/06, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi ! I always use your ARI through AAH, and indeed nice job ! A few comment : - I have seen that we could use ARI only for the Call Monitor by setting a value. would it be possible to do the same for only Voicemail ... indeed, we are using Asterisk only for Voicemail, and this would be so good only to present this tab to people ... ( And in Settings page also, hiding the Call Monitor Settings part here too ) - Same for help ( to show it or not ) I have installed it on our AAH 1.3 version and here are the error messages I get : Call Monitor Page (Only the first message on each page shows the Play link): Warning: is_dir(): Stat failed for /var/lib/asterisk/bin/archive_recordings/ (errno=20 - Not a directory) in /var/www/html/recordings/includes/bootstrap.inc on line 113 Settings Page (Didn't try to apply new settings): Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 434 Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 473 Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 577 I hope you won't take these comments as critics, you are really doing a GREAT job ! Asterisk was really lacking this application part ! Thanks again, And all the best ! Jean-Marc On 2/17/06, Dan Littlejohn [EMAIL PROTECTED] wrote: ARI(Asterisk Recording Interface) has reached another milestone. The project is starting to become a full featured user portal and handle all the common errors that people seem to have.This release supports: call monitor page – new features include column sorting and filter small duration calls in addition to the ability to listen to call monitor recordings voicemail page – allows voicemail message listening and management handset feature code help page - I can never remember them all user settings web interface - that allows setting call fowarding, voicemail email and pager, voicemail password, and call monitor recording There are also alot of i18n translations now, although with all the rework of the code many are now somewhat broken and need to be updated.If you speak one of the following, email and I will send you the page to translate or updating to the appropriate ari.po page and returning it to me would be very helpful. German Greek Spanish French Hebrew Hungarian Italian Portuguese Swedish If you would like to translate ARI into another language, I would be happy to support it. Loaded into AMP CVS and also here: www.littlejohnconsulting.com?q=ari If you have a chance, take a look.Comments and suggestions are welcome. Dan 512.791.0137 www.littlejohnconsulting.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 Jean-Marc:Thanks for the feedback.I have addressed these issues they areavailable on my website and have been checked into AMP cvs. I have added a setting to the /recording/includes/main.conf file.$ARI_DISABLED_MODULES = ; allows forindividual modules to be disabled (they are truemodules though, and you can just delete them from the /recordings/modules directory)the is_dir error is a PHP bug. http://groups.google.com/group/mailing.www.php-dev/browse_frm/thread/1b5b94e775b70cdb/877e4406600a8121?lnk=stq=Warning%3A+is_dir()%3A+Stat+failed+for+errno%3D20+-+Not+a+directoryrnum=1hl=en#877e4406600a8121 But, I think I was able to suppress the error.The settings page errors have been corrected.Thanks;Dan512.791.0137www.littlejohnconsulting.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] ARI 0.06
Hi ! I always use your ARI through AAH, and indeed nice job ! A few comment : - I have seen that we could use ARIonly for the Call Monitor by setting a value. would it be possible to do the same for only Voicemail ... indeed, we are using Asterisk only for Voicemail, and this would be so good only to present this tab to people ... ( And in Settings page also, hiding the Call Monitor Settings part here too ) - Same forhelp ( to show it or not ) I have installed it on our AAH 1.3 version and here are the error messages I get : Call Monitor Page (Only the first message on each page shows the Play link):Warning: is_dir(): Stat failed for /var/lib/asterisk/bin/archive_recordings/ (errno=20 - Not a directory) in /var/www/html/recordings/includes/bootstrap.inc on line 113 Settings Page (Didn't try to apply new settings):Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 434Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 473 Warning: Invalid argument supplied for foreach() in /var/www/html/recordings/modules/settings.module on line 577 Ihope you won't take these comments as critics,you are really doing a GREAT job ! Asterisk was really lacking this application part ! Thanks again, And all the best ! Jean-Marc On 2/17/06, Dan Littlejohn [EMAIL PROTECTED] wrote: ARI(Asterisk Recording Interface) has reached another milestone.The project is starting to become a full featured user portal and handle all the common errors that people seem to have.This releasesupports:call monitor page – new features include column sorting and filtersmall duration calls in addition to the ability to listen to call monitor recordingsvoicemail page – allows voicemail message listening and managementhandset feature code help page - I can never remember them alluser settings web interface - that allows setting call fowarding, voicemail email and pager, voicemailpassword, and call monitor recordingThere are also alot of i18n translations now, although with all therework of the code many are now somewhat broken and need to be updated.If you speak one of the following, email and I will send youthe page to translate or updating to the appropriate ari.po page andreturning it to me would be very helpful.GermanGreekSpanish FrenchHebrewHungarianItalianPortugueseSwedishIf you would like to translate ARI into another language, I would behappy to support it.Loaded into AMP CVS and also here: www.littlejohnconsulting.com?q=ariIf you have a chance, take a look.Comments and suggestions are welcome.Dan512.791.0137www.littlejohnconsulting.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] SIP Header VIA when behind NAT
I 'm wondering ... I have tried to use Asterisk external IP for some times ... but it never affects the VIA SIP Field Is It normal ? When reading many books in SIP, this should be external IP, no ? Did somebody manage to put/force the external IP in this VIA header ? If yes, how ? If not, any ideas how to reach this goal? Thanks, JM ___ --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] How to Get SIP Header : To Field ?
Hi,
I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch.
When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk :In INVITE : Vm Phone Number ( to route the call )In To : Person who has been called !In From : Person who was calling !
Of course, I need to send the call into the Called User Mailbox (Thus To SIP header) !
So Basically, filed in INVITE is EXTEN, From field can be obtained from the function ${SIPCHANINFO(from)}But how to get the To field ?
I have tried to add some code line into the chan_sip.c ...It works partially ... meaning that, I can add this to in SIPCHANINFO funciton,but the result is null.
Here is what I have added in chan_sip.c :in structure sip_pvt ( to field same as from )in sipchaninfo_function added to Line same as fromfunction_sipchaninfo_read added to line same as from
So I believe that I have enabled somehow Asterisk to read the value to from the channel ...But how to get the value and put it inside the channel ??? I think this would be my real question !
Thanks in advance for anybody who could help me ...
Yours,
JM
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[Asterisk-Users] Using Conferencing and Meetme
Hi, I ve installed recently AAH 1.1 And I was wondering on how to use this conferencing feature ? I have created extension 200. and when I try to call 8200, it says that this is not a valid conference number. Is there something specific to do ? Also, when entering MeetMe console, I cannot see anything. Is that allright ? meaning that if I have not started any conferencing, then, I shall see nothing in MeetMe :o) Thanks for any help ! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Conferencing and Meetme
I did that ... 8200 ... still ... nothing ... :o) On 6/28/05, Dean Collins [EMAIL PROTECTED] wrote: Also, when entering MeetMe console, I cannot see anything. Is that allright ? meaning that if I have not started any conferencing, then, I shall see nothing in MeetMe :o) You need to type the extension number in the box to see the conference details ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
