RE: [Asterisk-Users] WebVMail Woirks but No Audio
I had this issue- it's security on the files. I put a cron job that do /bin/chmod 777 /var/spool/asterisk/voicemail/default -R evey 1 minute, but there may be a cleaner solution. Assaf Benharoosh MCP, MCSA, MCSE [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard J. Sears Sent: Friday, February 25, 2005 11:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] WebVMail Woirks but No Audio Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] GET /vmail/vmail.cgi?action=audiofolder=INBOXmailbox=2377context=default password=12msgid=format=gsmdontcasheme=4624.gsm HTTP/1.1 200 9438 - contype But the box at the bottom shows up as a broken link. Any ideas...? Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ 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 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] Channel Variable
Bill,
Are
you sure there's an AGI enviroment variable that gives me that? I couldn't find
any:
-- accountcode =-- callerid = "Assaf Benharoosh"
21-- channel = SIP/26-f39a-- context =
extensions-- dnid = 45-- enhanced = 0.0--
extension = 45-- language = en-- priority = 1--
rdnis = unknown-- request = agi-test.agi-- type =
SIP-- uniqueid = 1105227054.140
I'm
trying to get the other side channel string. If I run the AGI before the dial-
is it all possible?
Assaf BenharooshMCP, MCSA, MCSE[EMAIL PROTECTED]Frantic,
LLC.246 West 38th
Street2nd FloorNew York, NY
10018T: (212)
302-5790F: (646) 201-9418C: (516) 805-7981
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
SeddonSent: Saturday, January 08, 2005 4:21 AMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] Channel Variable
Does anyone know how to get
the channel ID on the other side of the call?
Assaf, I dont know if
there is such an ID available. However if there is not, the value you want
is pushed out in one of the events that Asterisk publishes to AGI connections
when a call is constructed. As it result it ought to be possible to write
an AGI script using, say, Perl to capture this value and write it back as a
Dialplan variable.
Bill
Seddon
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Assaf BenharooshSent: January 08, 2005 12:27 AMTo: Asterisk Users Mailing List -
Non-Commercial DiscussionSubject: [Asterisk-Users] Channel
Variable
Hi all,
Does anyone know how to get the
channel ID on the other side of the call?
For example: When SIP/50 calls
SIP/21, and the call is answered by SIP/21 I get:
SIP/21-6735 answered
SIP/50-b456
${CHANNEL} will show me SIP/50-b456.
Is there a parameter or a workaround
to get the SIP/21-6735 part?
Thanks.
Assaf
Benharoosh
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[Asterisk-Users] Channel Variable
Hi
all,
Does anyone know how
to get the channel ID on the other side of the call?
For example: When
SIP/50 calls SIP/21, and the call is answered by SIP/21 I
get:
SIP/21-6735 answered
SIP/50-b456
${CHANNEL} will show
me SIP/50-b456.
Is there a parameter
or a workaround to get the SIP/21-6735 part?
Thanks.
Assaf Benharoosh
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[Asterisk-Users] HDLC
Hello, Does anyone have experience in HDLC setup with T100P ? I've looked around, including wiki pages, and still having problem in the ztcfg stage (or even in the make stage). Do I need to do something in the kernel for the HDLC first? I'm using 2.4.20 kernel. Thanks, Assaf Benharoosh ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] CNG Comfort Noise Generation
Thank you for making this clear for me. Is there any solution for the mentioned phones? Assaf Benharoosh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Saturday, November 13, 2004 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CNG Comfort Noise Generation Hi Assaf, Assaf Benharoosh wrote: I have a problem with many phone such as BudgeTone, ariaVoice, PCPhoneline. They are not generating comfort noise (you can hear yourself when you're talking)- with budgetone having CNG sporadically. Is there a way to make this happen on Asterisk - or it must be a phone feature. Does anyone else experiencing this issue with those phones and have a workaround? Assaf Benharoosh Hearing yourself when you talk is not comfort noise. It is sidetone. Comfort noise is simulating the background noise of the room at the far end when nobody is talking and transmission has stopped. Sidetone is always a phone feature. Comfort noise usually is too. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CNG Comfort Noise Generation
I have a problem with many phone such as BudgeTone, ariaVoice, PCPhoneline. They are not generating comfort noise (you can hear yourself when you're talking)- with budgetone having CNG sporadically. Is there a way to make this happen on Asterisk - or it must be a phone feature. Does anyone else experiencing this issue with those phones and have a workaround? Assaf Benharoosh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] C in Dial doesn't work (no cdr)
Has anyone experienced this problem? The C flag in Dial app doesn't work. I'm getting the CDR record although it has C (reset CDR for this call). The C is even recorded in the CDR record lastdata field. Assaf BenharooshMCP, MCSA, MCSE[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 to Vonage
I made it work: Sip.conf: register = 1yournumber:secret@atlas-east.vonage.net:5060 [vonage] type=friend username=1yournumber secret=secret host=atlas-east.vonage.net port=5060 allow=all maxexpirey=15 dtmfmode=inband fromuser=1yournumber fromdomain=atlas-east.vonage.net canreinvite=no nat=yes context=yourcontext Extensions.conf: exten = _1yournumber,1,your app Enjoy Assaf. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paterson, Mark Sent: Tuesday, August 24, 2004 7:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk to Vonage Honestly, do you think I would ask for help on the list if I hadn't come up with any successful results on my own?? Just asking if anyone has made this work. If so what rev of * were they running and what do their configs look like. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Tuesday, August 24, 2004 6:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk to Vonage Yes, search google for asterisk vonage working site:lists.digium.com -Original Message- From: Paterson, Mark [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 11:19 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk to Vonage I'm trying to connect my Asterisk server via sip using my vonage soft phone account. Has any anyone successfully got to work? I get error from asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: Unable to lookup '216.115.25.199:5061' when trying to register with the vonage sip proxy. Any examples would be greatly appreciated. Rgs, mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IP Phone- disjoin conference
Hi all, Does anyone know if it's possible to continue talking to one member of an initiated conference call on Cisco 79xx ? In other words- disconnect one of the parties. Thanks. Assaf
RE: [Asterisk-Users] Vonage working with asterisk
I still didn't get it to work. When calling the number- it goes to voicemail. No indication on the CLI. The 'sip show peers' shows: vonage/16464855 216.115.25.199 N 255.255.255.255 5061 Unmonitored 'sip show registry': HostUsername Refresh State sphone.vopr.vonage.net:5061 16464855183 15 Registered Help anyone? Assaf Benharoosh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 14, 2004 6:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vonage working with asterisk atlast after working of 7 hours i got voange soft account working on asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 and Linejacks
Hi, I actually gave up on the LineJack. I'm using Digium 4 FXO card- which does the job pretty well. Assaf Benharoosh MCP, MCSA, MCSE [EMAIL PROTECTED] Frantic, LLC. 246 West 38th Street 2nd Floor New York, NY 10018 T: (212) 302-5790 F: (646) 201-9418 C: (516) 805-7981 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of greg Sent: Thursday, July 22, 2004 2:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Linejacks I found a message from you to the asterisk users mailing list from 2001. I was wondering if you got (or still have) an asterisk system working with the linejack? If so, would you be willing to assist me with mine? I seem to have things working, and * says that caller ID is coming in, but I can't get * to actually answer the call. Thanks, Greg -- NetIO.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voicepulse Down Again?
I've been getting the same type of answers for the past month. Assaf Benharoosh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, June 16, 2004 12:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicepulse Down Again? Here is the mail I just received from VoicePulse Hello Steve, Thank you for contacting VoicePulse. The issue with VoicePulse Connect! has been resolved. Please verify that Connect! is working. Our engineers are working to add more servers in the next few days to handle the increased call volume. Please reply directly to this email if we can provide any additional assistance. Regards, VoicePulse Customer Support -- VoicePulse Now Offers Unlimited Calling for $24.99 See more info at: http://www.voicepulse.com/plans/ -- Find the answers to your most common questions at: http://www.voicepulse.com/kb -- On Wed, 16 Jun 2004 11:12:38 -0400, [EMAIL PROTECTED] wrote: Hello, I have been having intermittent problems with registering over iax to Voicepulse Connect. Are you aware of any problems? I have published this number to customers. Thanks, Steve Totaro Totaro Technologies, Inc. - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 15, 2004 10:05 PM Subject: Re: [Asterisk-Users] Voicepulse Down Again? It was down and now its back up. For all the future Asterisk Users out there that will read this post in the archives. Voicepulse has had intermittant problems. Keep that in mind while you are shopping for IAX2 (proprietary to * users) - Original Message - From: twisted [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 15, 2004 9:46 PM Subject: Re: [Asterisk-Users] Voicepulse Down Again? On Tue, 2004-06-15 at 19:20, Steve Totaro wrote: Then where is? A good many Asterisk Users use voicepulse connect so I would say it does. sgt - Original Message - From: Brian K. West To: [EMAIL PROTECTED] Sent: Tuesday, June 15, 2004 8:08 PM Subject: Re: [Asterisk-Users] Voicepulse Down Again? This is not the place for these typs of messages. bkw See my post from last night about etiquette. IRC is one example of somewhere you can ask this question and it would be acceptable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] VoicePulse broken?
They had about 5 down times today- between 2 hours and 20 minutes each :-( Assaf Benharoosh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David H Hickman Sent: Thursday, May 20, 2004 3:13 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse broken? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 They are still down as of 1415CST. I believe that I am using gw5. dhh On May 20, 2004, at 1:45 PM, Zac Amsler wrote: They had an issue this morning. It is fixed now. Zac -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Weis Sent: Thursday, May 20, 2004 1:01 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse broken? Inbound is working here, no problems that I know of. Scott - Original Message - From: C. Sullivan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 20, 2004 12:52 PM Subject: [Asterisk-Users] VoicePulse broken? Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFArQNeWzu/tX8BYR8RAqKlAJwJSKhQ7X2+YkYAvLxq61kDg1C36QCfUbHP l59A4WnmMLGAXoReVrIQAzc= =ey0i -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse 1-800 numbers sound problem
To whom it may concern, When dialing out an 800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This is not due to a problem on your Asterisk or your line- the bad soundeffect occurs in VoicePulse. (just spend lots of time finding that out) Assaf BenharooshMCP, MCSA, MCSE[EMAIL PROTECTED]Frantic, LLC.
[Asterisk-Users] VoicePulse 1-800 numbers sound problem
Sorry- wrong observation. The problem is when placing a call to IAX from a Cisco 7940. To whom it may concern, When dialing out an 800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This is not due to a problem on your Asterisk or your line- the bad soundeffect occurs in VoicePulse. (just spend lots of time finding that out) Assaf BenharooshMCP, MCSA, MCSE[EMAIL PROTECTED]Frantic, LLC.
