RE: [Asterisk-Users] Voicemail limit?
Hi, I'm using version 1.0.9. In the sip.conf or any configuration file, is there any parameter that specify the size of users? Thanks, Ryan At 03:21 AM 3/28/2006, Watkins, Bradley wrote: What version of Asterisk are you running? The site that I have with that many users is 1.2.0 (plus a couple of custom patches, but nothing to app_voicemail.c that would make it more scalable). Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil Sent: Sunday, March 26, 2006 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicemail limit? Hi Brad, On my sip.conf I have 83 users also in my voicemail.conf but when I call the users above 70 it prompts me User Not Found. Any idea regarding this? Thanks, Ryan At 04:12 AM 3/24/2006, Watkins, Bradley wrote: I don't think there's any kind of (significantly small, anyway) limit. I have over 300 users at one site in voicemail.conf and no issues there. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil Sent: Wednesday, March 22, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail limit? Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 limit?
Hi Brad, On my sip.conf I have 83 users also in my voicemail.conf but when I call the users above 70 it prompts me User Not Found. Any idea regarding this? Thanks, Ryan At 04:12 AM 3/24/2006, Watkins, Bradley wrote: I don't think there's any kind of (significantly small, anyway) limit. I have over 300 users at one site in voicemail.conf and no issues there. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil Sent: Wednesday, March 22, 2006 2:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail limit? Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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 limit?
Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan ___ --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 limit?
Hi, Is there a howto to do this? I'm using voicemail.conf and sip.conf for my voicemail users. Does it really has a limit? Thanks, Ryan At 08:23 PM 3/23/2006, Antonio Rabena wrote: How about moving your voicemail users into db? At 03:50 AM 3/23/2006, you wrote: Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan ___ --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 limit?
Hi Antonio, Thanks for the reply and for the links. Regards, Ryan At 09:14 PM 3/23/2006, Antonio Rabena wrote: You can try using asterisk-addons http://www.voip-info.org/wiki/view/Asterisk+voicemail+database or asterisk realtime http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail At 04:51 AM 3/23/2006, you wrote: Hi, Is there a howto to do this? I'm using voicemail.conf and sip.conf for my voicemail users. Does it really has a limit? Thanks, Ryan At 08:23 PM 3/23/2006, Antonio Rabena wrote: How about moving your voicemail users into db? At 03:50 AM 3/23/2006, you wrote: Hi, Is there an account limit for voicemail? I have 80+ users in the voicemail and I can only reach the 70-ieth user. If there is a limit how can I increase it to hundred for example? Thanks, Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Registering with SER question
Hi Olle, Nice to know that. In my case I'm simulating a prepaid call from Asterisk to SER. On the Asterisk side, there are users registered with of course different extensions. Asterisk uses SER as the SIP trunk and SER will forward it to the PSTN gateway. Asterisk registers to SER with single username asterisk, and assuming that the asterisk-registered-users placed calls simultateously, on my CDR database there will be multiple occurence of the asterisk username because of multiple calls. Now if total duration of all the calls placed by username asterisk greater than his credit, I will send a BYE to them for them to disconnect. Would it be usefull not setting up the extension on my register= parameter in sip.conf? If I used the sip:[EMAIL PROTECTED] will it distuinguish the asterisk registered user to disconnect? I'm using a perl script for me to monitor the calls on SER, also sipsak. Thanks, Ryan At 03:28 PM 2/1/06, Olle E Johansson wrote: Ryan Pagquil wrote: Hi, On asterisk console I enabled SIP debugging and I found out that asterisk is sending this: Reliably Transmitting: REGISTER sip:imydomain.com SIP/2.0 Via: SIP/2.0/UDP :x.x.x.x:5060;branch=z9hG4bK69398d1a From: sip:[EMAIL PROTECTED];tag=as1d1a85bc To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 *Contact: sip:[EMAIL PROTECTED] --registered on SER Contact column on location table *Event: registration Content-Length: 0 so it means that Asterisk is sending that information, how can I correct this? It should be sip:[EMAIL PROTECTED] no sip:[EMAIL PROTECTED] . Ryan, Check the syntax of your register= statement. The last entry is the extension. If you are not entering any extension, asterisk will send s as in this case. You have plenty of examples in sip.conf.sample in the /configs directory of your source code. /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 ___ --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 Registering with SER question
Hi, I've been registering asterisk to ser. I'm using SER as the outbound SIP trunk for Asterisk. Users registered with Asterisk will use the SIP trunk to reach SER registered users and PSTN's. Now when I register Asterisk with SER, on my SER's location table I see these record: Username Column = asterisk Contact Column = sip:[EMAIL PROTECTED] I have a script running that checks the accounting records and sends BYE for the username that has no credit left. I found it hard doing this because of the record on the Contact column on location table of SER, everytime asterisk registers with SER. I could not send BYE to asterisk because of the broken contact information on Contact column of SER's location table. How can I correct this? Here is my sip.conf configuration: [general] port = 5060 bindaddr = x.x.x.x context = sip disallow=all allow=ulaw allow=alow fromuser=asterisk secret=test123 realm=mydomain.com register=asterisk:[EMAIL PROTECTED] Thanks in advance, Ryan ___ --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 Registering with SER question
Hi, On asterisk console I enabled SIP debugging and I found out that asterisk is sending this: Reliably Transmitting: REGISTER sip:imydomain.com SIP/2.0 Via: SIP/2.0/UDP :x.x.x.x:5060;branch=z9hG4bK69398d1a From: sip:[EMAIL PROTECTED];tag=as1d1a85bc To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] --registered on SER Contact column on location table Event: registration Content-Length: 0 so it means that Asterisk is sending that information, how can I correct this? It should be sip:[EMAIL PROTECTED] no sip:[EMAIL PROTECTED] . Thanks in advance, Ryan ___ --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 Registering with SER question
Hi, What does it means? Thanks, Ryan At 01:40 PM 2/2/06, Abhishek wrote: Hi ryan , The header you are suspecting does not contains the registration info. , it is actually the return path for the ACK which will get generated in response to this packet. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ryan Pagquil Sent: Tuesday, January 31, 2006 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Registering with SER question Hi, On asterisk console I enabled SIP debugging and I found out that asterisk is sending this: Reliably Transmitting: REGISTER sip:imydomain.com SIP/2.0 Via: SIP/2.0/UDP :x.x.x.x:5060;branch=z9hG4bK69398d1a From: sip:[EMAIL PROTECTED];tag=as1d1a85bc To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] --registered on SER Contact column on location table Event: registration Content-Length: 0 so it means that Asterisk is sending that information, how can I correct this? It should be sip:[EMAIL PROTECTED] no sip:[EMAIL PROTECTED] . Thanks in advance, Ryan ___ --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] SIP Trunk please help
Hi,
I already contacted what I inputed on my softphone but we
both can't hear each other. I used X-lite and the other is a hardware
SIP phone. What could be the problem?
Thanks,
Ryan
At 03:03 PM 12/16/05, you wrote:
yes
$AGI-exec('Dial', SIP/[EMAIL PROTECTED]);
Diyanat
From: Ryan Pagquil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com, asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 13:56:09 +0800
MIME-Version: 1.0
X-OriginalArrivalTime: 16 Dec 2005 05:58:00.0170 (UTC)
FILETIME=[AB7B14A0:01C60205]
Hi,
Thanks for the reply... Actually I'm using AGI to do it
instead of defining it on extensions.conf... Would it be the same
in extensions.conf? Should I write $AGI-exec('Dial',
'SIP/[EMAIL PROTECTED]'); to dial it from AGI script (perl), is this correct?
Thank you very much,
Ryan
At 01:45 PM 12/16/05, Diyanat Ali wrote:
in the sip.conf have the following enteries
; for regsitering with ser
register:seruser:[EMAIL PROTECTED]:5060;(put ser machine ip:port)
;add a user for the ser machine
[seruser]
type=friend
host=0.0.0.0 ;(put ser machine ip here)
nat=no ;(change as needed )
canreinvite=yes ;(change as needed)
insecure=very ;(change as needed)
disallow=all
allow=ulaw
allow=gsm
context=sip
dtmfmode=rfc2833
in extensions.conf under contect [sip]
[sip]
;replace extension and the priority to macth your dial plan
exten = _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ;(seruser is defined
in sip.conf)
Diyanat
From: Ryan Pagquil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 10:31:24 +0800
MIME-Version: 1.0
Hi,
I've been setting up asterisk for prepaid use. I'm
testing to call a SER registered user from the Asterisk just to
simulate the prepaid calls. Now, I can already contact Asterisk
and it prompts me to input my call card number and after that I
dial in the number I want to call (a SER registered device). My
question is how can I implement on sip.conf to use my SER as the
trunk line? So that calls will be forwarded to it. Do I also
need to register asterisk on SER?How?
Please help!
Thanks,
Ryan
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[Asterisk-Users] SIP Trunk please help
Hi, I've been setting up asterisk for prepaid use. I'm testing to call a SER registered user from the Asterisk just to simulate the prepaid calls. Now, I can already contact Asterisk and it prompts me to input my call card number and after that I dial in the number I want to call (a SER registered device). My question is how can I implement on sip.conf to use my SER as the trunk line? So that calls will be forwarded to it. Do I also need to register asterisk on SER?How? Please help! Thanks, Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Trunk please help
Hi,
Thanks for the reply... Actually I'm using AGI to do it
instead of defining it on extensions.conf... Would it be the same in
extensions.conf? Should I write $AGI-exec('Dial',
'SIP/[EMAIL PROTECTED]'); to dial it from AGI script (perl), is this correct?
Thank you very much,
Ryan
At 01:45 PM 12/16/05, Diyanat Ali wrote:
in the sip.conf have the following enteries
; for regsitering with ser
register:seruser:[EMAIL PROTECTED]:5060;(put ser machine ip:port)
;add a user for the ser machine
[seruser]
type=friend
host=0.0.0.0 ;(put ser machine ip here)
nat=no ;(change as needed )
canreinvite=yes ;(change as needed)
insecure=very ;(change as needed)
disallow=all
allow=ulaw
allow=gsm
context=sip
dtmfmode=rfc2833
in extensions.conf under contect [sip]
[sip]
;replace extension and the priority to macth your dial plan
exten = _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ;(seruser is defined in
sip.conf)
Diyanat
From: Ryan Pagquil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 10:31:24 +0800
MIME-Version: 1.0
Hi,
I've been setting up asterisk for prepaid use. I'm testing
to call a SER registered user from the Asterisk just to simulate
the prepaid calls. Now, I can already contact Asterisk and it
prompts me to input my call card number and after that I dial in
the number I want to call (a SER registered device). My question
is how can I implement on sip.conf to use my SER as the trunk
line? So that calls will be forwarded to it. Do I also need to
register asterisk on SER?How?
Please help!
Thanks,
Ryan
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[Asterisk-Users] Please help in writing AGI script
Hi,
I'm new in writing AGI script and actually newbie in Asterisk. I'm
writing a small script that will read the number inputed by the
caller of the extension 123. First he will dial number 123 then a
voice prompt will be played (welcome) then he should press number on
the softphone and the script will echo the number to the caller. Here
is my script:
#!/usr/bin/perl
use Asterisk::AGI;
$|=1;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-stream_file('welcome');
while(length($num) != 3) {
$num = $AGI-get_data(sayme, 1, 3);
$saythis = $num;
}
$AGI-say_number($saythis);
please correct my script if there is something wrong. but i think there is.
thanks,
ryan
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[Asterisk-Users] Voicemail email format, please help!
Hi, I'm now using Asterisk for my voicemail together with SER. They just work fine. When the user in SER is not registered the call will be forwarded to Asterisk and the caller will record his message. Then I also made asterisk to send the wav as attachment to its email. I try using two ip phones one is Xlite and the other is a hardware ip phone to call the voicemail. When asterisk sent the mail to me I found that the voicemail from the hardware ip phone has the display username and the number of the caller, but the Xlite voicemail only has the display username... then I checked the voicemail box of my username and check the message text that corresponds to the voicemail and found these: hardware ip phone: [message] origmailbox=810020 context=ser macrocontext= exten=u810020 priority=1 callerchan=SIP/mydomain.com-0018b368 callerid=test3 103 origdate=Thu Nov 17 11:22:09 AM GMT 2005 origtime=1132226529 duration=31 xlite: [message] origmailbox=810020 context=ser macrocontext= exten=u810020 priority=1 callerchan=SIP/810020-e30c callerid=810020 origdate=Thu Nov 17 11:21:38 AM GMT 2005 origtime=1132226498 duration=31 How come does the hardware phone has the 103 on the callerid and xlite don't have its number? Is this a misconfiguration? Thanks, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel compilation help!
Hi, I'm compiling Zaptel1-1.0.9 in Sparc64/Debian and I'm getting these errors. I compiled asterisk on the same machine and it went ok. I want to activate the conference feature of asterisk thats why i'm compiling zaptel. These are the errors: sip:/usr/local/src/zaptel-1.0.9.1# make gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DSTANDALONE_ZAPATA -c zaptel.c In file included from /usr/include/linux/dcache.h:10, from /usr/include/linux/fs.h:17, from /usr/include/linux/proc_fs.h:6, from zaptel.c:45: /usr/include/linux/rcupdate.h: In function `rcu_pending': /usr/include/linux/rcupdate.h:114: error: invalid lvalue in unary `' /usr/include/linux/rcupdate.h:116: error: invalid lvalue in unary `' /usr/include/linux/rcupdate.h:117: error: invalid lvalue in unary `' zaptel.c: In function `zt_register': zaptel.c:4406: warning: implicit declaration of function `class_simple_device_add' zaptel.c: In function `zt_unregister': zaptel.c:4456: warning: implicit declaration of function `class_simple_device_remove' zaptel.c: In function `zt_init': zaptel.c:6431: warning: implicit declaration of function `class_simple_create' zaptel.c:6431: warning: assignment makes pointer from integer without a cast zaptel.c: In function `zt_cleanup': zaptel.c:6492: warning: implicit declaration of function `class_simple_destroy' make: *** [zaptel.o] Error 1 please help, Thanks, Ryan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recording voice messages in mp3 format
Hi, Is there a way so that I can record the voice messages in mp3 format instead of wav? I think it is much smaller in size compare to wav. It is also easier to send small sized file as an attachment. Currently when my users record voice messages the format is wav. Where can I configure it so that it will become mp3? Thanks, Ryan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording voice messages in mp3 format
Hi, Yes, I'm using wav for my recording and the file is quite large. Ryan At 04:43 PM 11/16/05, Gerard Dupont III wrote: Are you using wav or wav49? You can check in /etc/asterisk/voicemail.conf under the format option... wav49 creates much smaller files than normal wav and doesn't need a special player like gsm files would and as far as using mp3, I'm not sure how to go about that. -Gerard Ryan Pagquil wrote: Hi, Is there a way so that I can record the voice messages in mp3 format instead of wav? I think it is much smaller in size compare to wav. It is also easier to send small sized file as an attachment. Currently when my users record voice messages the format is wav. Where can I configure it so that it will become mp3? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk: BUS Error in SPARC/Linux (debian)
Hi, I used Aterisk 1.0.7 Debian package and it is now running. I noticed that when I used X-lite and GSM codec is enabled Aterisk stops and says Bus Error. Here is the message i got: *CLI -- Executing VoiceMailMain(SIP/mydomain.com-0011a700, 810020) in new stack -- Playing 'vm-login' (language 'en') Bus error But when I disable GSM on X-lite Asterisk doesn't hang. What seems to be the problem here? Is there an issue regarding GSM on Debian/Sparc64? I think same thing happened when I first used Asterisk-1.0.9 version. Please help... Thanks, Ryan At 02:02 PM 11/13/05, Tzafrir Cohen wrote: On Fri, Nov 11, 2005 at 11:08:25AM +0800, Ryan Pagquil wrote: Hi, I successfully installed asterisk in SPARC64/Linux as the voicemail for my SER installation. No problem when I run it, but the problem is when I forward the voicemail traffic to it, on the user agent side I heard the start of the voice prompt but immediately stopped. I then checked on the server and it says Bus error. What can I do to fix this? Which version of Asterisk? of Debian? If this is from an official deb. maybe you should use reportbug(1). -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk: BUS Error in SPARC/Linux (debian)
Hi, I successfully installed asterisk in SPARC64/Linux as the voicemail for my SER installation. No problem when I run it, but the problem is when I forward the voicemail traffic to it, on the user agent side I heard the start of the voice prompt but immediately stopped. I then checked on the server and it says Bus error. What can I do to fix this? Thanks in advance, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I got 403, Forbidden... please help
Hi Harry, I tried your suggestion and it worked. But I don't hear any voice from the anonymous user. I don't hear the voice prompt? What should I do? Thanks, Ryan harry gaillac wrote: Hello, Try insecure=very in [sip.philonline.com] Harry --- Ryan Pagquil [EMAIL PROTECTED] a écrit : Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts 403, Forbidden . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I need to do to make those anonymous callers to reach the voicemails of my users? here is my sip.conf. [general] port = 5060 bindaddr = 202.84.24.47 context = sip disallow=all allow=ulaw allow=alow ;register=me:[EMAIL PROTECTED]/1000 [sip.philonline.com] type=friend host=sip.philonline.com fromuser=rpagquil secret=test123 fromdomain=sip.philonline.com [phone1] type = friend username = phone1 secret = test123 host = dynamic context = sip mailbox = callerid=Test1 [acjeff] type=friend username=acjeff host=dynamic defaultip=10.0.1.236 nat=yes context=sip mailbox= callerid=Test2 [usser1] type = friend username = usser1 secret = test123 nat=yes host = dynamic context = sip mailbox = 111 callerid=User1 Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to retrieve voicemail from an IP phone?
Hi, How can I retrieve those voicemails using my ip phone? and how will i confiugre it on asterisk? Please help I'm very new in asterisk. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?
Hi Rich,
Does the user need to dial his extension just to retrieve the
voicemails or he will dial other number to access those voicemails?
In the config does it mean that when a user dial 3998 he will be able to
retrieve those voicemails? So it means that every users must have a
mailbox number for which they will retrive their voicemails? I'm really
a newbie. =)
Thanks fo the help,
--ryan
Rich Adamson wrote:
How can I retrieve those voicemails using my ip phone? and how
will i confiugre it on asterisk?
Please help I'm very new in asterisk.
Add something like this in your extensions.conf file:
; Voicemail access (prompts for exten and password)
exten = 3998,1,Wait,1
exten = 3998,2,VoicemailMain
exten = 3998,3,Hangup
; Voicemail access (does not prompt for anything)
exten = 3999,1,Wait,1
exten = 3999,2,VoicemailMain(s${CALLERIDNUM})
exten = 3999,3,Hangup
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--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
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Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?
Hi,
I already made working.
Thanks for the help,
Ryan
Rudolf Ladyzhenskii wrote:
Hi,
You need a single extension to call voicemail. I am using 100.
extensions.conf
exten =100,1,VoiceMailMain(${CALLERIDNUM})
exten =100,2,Hangup()
Now, if you simply call VoiceMailMain() without parameters, voicemail
system will ask you to enter the number of mailbox you want to access.
This is useful if you want to read any mailbox from any phone.
However, if you specify a parameter like I did, voicemail will
automatically go into mailbox for the extension you have called from.
There is a little trick to get it work, though. Normally caller ID is
a name like Joe Smith
You will have to specify caller ID per user like that: (sip.conf for
example)
[user1]
callerid=Joe Smith 101
This will present asterisk with a way to get both name and extension
number.
Rudolf
- Original Message - From: Ryan Pagquil
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 21, 2005 8:58 PM
Subject: Re: [Asterisk-Users] How to retrieve voicemail from an IP phone?
Hi Rich,
Does the user need to dial his extension just to retrieve the
voicemails or he will dial other number to access those voicemails?
In the config does it mean that when a user dial 3998 he will be able
to retrieve those voicemails? So it means that every users must have
a mailbox number for which they will retrive their voicemails? I'm
really a newbie. =)
Thanks fo the help,
--ryan
Rich Adamson wrote:
How can I retrieve those voicemails using my ip phone? and
how will i confiugre it on asterisk?
Please help I'm very new in asterisk.
Add something like this in your extensions.conf file:
; Voicemail access (prompts for exten and password)
exten = 3998,1,Wait,1
exten = 3998,2,VoicemailMain
exten = 3998,3,Hangup
; Voicemail access (does not prompt for anything)
exten = 3999,1,Wait,1
exten = 3999,2,VoicemailMain(s${CALLERIDNUM})
exten = 3999,3,Hangup
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--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
___
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To UNSUBSCRIBE or update options visit:
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--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
___
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[Asterisk-Users] I got 403, Forbidden... please help
Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts 403, Forbidden . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I need to do to make those anonymous callers to reach the voicemails of my users? here is my sip.conf. [general] port = 5060 bindaddr = 202.84.24.47 context = sip disallow=all allow=ulaw allow=alow ;register=me:[EMAIL PROTECTED]/1000 [sip.philonline.com] type=friend host=sip.philonline.com fromuser=rpagquil secret=test123 fromdomain=sip.philonline.com [phone1] type = friend username = phone1 secret = test123 host = dynamic context = sip mailbox = callerid=Test1 [acjeff] type=friend username=acjeff host=dynamic defaultip=10.0.1.236 nat=yes context=sip mailbox= callerid=Test2 [usser1] type = friend username = usser1 secret = test123 nat=yes host = dynamic context = sip mailbox = 111 callerid=User1 Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
