[Asterisk-Users] No ring / Dead Air when transferring from our IVR

When we upgraded, our Asterisk system quit giving back a ring tone when transferring to an extension. It leaves dead air and the person calling thinks the phone went dead. Is there a setting or a wav file that I am needing? Todd Wallace

[Asterisk-Users] G.723

Is there an easy/cheap way to add g.723 to Asterisk? I have added g.729 and need g.723. Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

RE: [Asterisk-Users] G.723

Is it at all possible? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, April 12, 2004 12:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.723 Todd Wallace wrote: Is there an easy/cheap way to add g.723 to Asterisk

[Asterisk-Users] IP Phones that support G.723 on H.323

Does anyone know of Phone that supports G.723 on H.323. Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] H323 calls drop on connect

I have something new that is happening to me...When I call from a SIP phone and route out OH323, I get a good clear ringing, connect, then it drops me. If I get a telco recorded message, I hear the complete message. If I get a person that answers, I hear about the first 2 seconds, then it drops

RE: [Asterisk-Users] H323 calls drop on connect

] Behalf Of Todd Wallace Sent: Tuesday, March 02, 2004 4:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H323 calls drop on connect I have something new that is happening to me...When I call from a SIP phone and route out OH323, I get a good clear ringing, connect, then it drops me. If I get

RE: [Asterisk-Users] OH323 errors

version (asterisk-oh323-0.5.9)? This problem has been fixed. Michael. Todd Wallace wrote: I just pulled the latest OH323 and asterisk and compiled on Redhat 7.3 using Grandstream phones. I have a dial plan that goes out SIP when you hit a 9+number and H323 when you hit 7+number. SIP

RE: [Asterisk-Users] OH323 errors

-Users] OH323 errors Are you sure you are using the latest version (asterisk-oh323-0.5.9)? This problem has been fixed. Michael. Todd Wallace wrote: I just pulled the latest OH323 and asterisk and compiled on Redhat 7.3 using Grandstream phones. I have a dial plan that goes out SIP when you

[Asterisk-Users] Wrong Pitch 1st subfr.

I keep getting Wrong Pitch 1st subfr. when placing oh323 calls through asterisk. Not sure where to look as everything on oh323.conf looks fine. Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Buzzing on Grandstream phones

I get a low buzzing noise on my Grandstream phones when placing calls. Any one know how to get rid of that... Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] OH323 errors

I just pulled the latest OH323 and asterisk and compiled on Redhat 7.3 using Grandstream phones. I have a dial plan that goes out SIP when you hit a 9+number and H323 when you hit 7+number. SIP is very clean, but I get a scrolling Wrong Pitch 1st subfr. when you go out H323. Before, it was

[Asterisk-Users] Oh323 question

I have the following config: Asterisk compiled with oh323 on a public IP Grandstream behind a NAT Aseterisk sending calls to a Nextone MSW H.323 My Grandstream phone registers to my * server via SIP fine. When I place a call that goes from my * server to my Nextone

[Asterisk-Users] * with OH323 - Memory Leak

I noticed in the BUGS that there is a memory leak with * using asterisk-oh323. If we use SIP primarily as the main protocol, but OH323 on occasion to test some international routes on our Nextone MSW...How bad is the Memory leak that is described?? Todd Wallace

RE: [Asterisk-Users] Is it possible to push the media processing off to a gateway for processing?

to something with DSP's such as a gateway? That way, asterisk just has to handle the call setups and tear downs. Todd Wallace You mean, like what SIP does by default? This is an incomplete question. Please be more specific. If I have a gateway, and I have SIP calls coming in from desktop SIP UA's

[Asterisk-Users] Is it possible to push the media processing off to a gateway for processing?

I was wondering if it is possible to have Asterisk push the media processing off to something with DSP's such as a gateway? That way, asterisk just has to handle the call setups and tear downs. Todd Wallace ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] OH323

I had asterisk working with OH323 and it segmented faulted. Now when I send calls using the oh323 channel, it does not send IP address and I get blocked. Any thoughts? ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Notice Messages??? What does it mean

I just started getting the following notice message and was wondering what it meant. Jan 16 15:56:11 NOTICE[240654]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Todd Wallace

RE: [Asterisk-Users] Notice Messages??? What does it mean

Is there a way to tell which device it is coming from? We have Grandstream phones and it is intereconnected with a Nextone MSW using SIP. Todd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: Saturday, January 17, 2004 8:41 AM

[Asterisk-Users] call duration in the cdr

I have a question about the call duration and billable duration field. The end call time looks like it is 1 second off of the start time and does not match the call duration number. Any thoughts? Todd Wallace ___ Asterisk-Users mailing

[Asterisk-Users] detect third party

Caller 1 to be able to place a call to the third party and therefore want to detect the third party call. Is there a way to detect the flash that mom does? That flash is being supplied from the central office on the line side Is this possible??? Todd Wallace

RE: [Asterisk-Users] large implementation

I am also interested in large scale deployments. Todd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hauser Sent: Sunday, December 21, 2003 5:45 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] large implementation Hello, I am looking at

[Asterisk-Users] ringing

How do I turn off the initial ringing in Asterisk. I get an Euro sounding ringing prior the ringing from the carrier. I don't get it on the X100P, but do on the SIP outbound side. Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Budgetone phones

else is failing. Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] long delay on meetme

Is there a setting on the meetme room to shorten the delay. When someone speaks, there is a long delay until the sound is actually heard? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk

[Asterisk-Users] ringing

I get 2 ringing sounds when placing a SIP call through my carrier. the first sounds European for 1 ring then, it goes to a US ring. Any thoughts? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] ringing

Is there a wait or a setting that I can set so that * does not do this? It sounds like you're receiving ringback from your local asterisk first. Then, somewhere along the progress, your asterisk receives an open channel and connects you to the sip carrier. At this point, the carrier's channel

[Asterisk-Users] Web Page initiated phone to phone

Is it possible to initiate 2 outbound calls from a web page and conference them together in a bridge on an asterisk server? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] phone port on the x100p

Can the phone port on the x100p be an addressable extension on asterisk? I want to plug our conference phone into that phone jack as it is an analog phone. Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

[Asterisk-Users] DTMF

What DTMF options are available to me. My carrier is using DTMF relay H245 Alpha Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] More infor on my earlier DTMF question

, rfc2833, and inband. No luck. He has tried avail settings in the Nextone. We can't seem to sync up. I do not have this problem when dealing with the X100P, but I really want to have the call handed off SIP via this carrier. Anyone suggestions?? Todd Wallace

[Asterisk-Users] Asterisk and G.729a

Does asterisk support G.729a or do you have to add something (is there an open source one) Todd Wallace

[Asterisk-Users] Ringer on Grandstream Budetone 100 phone

Does anyone know if the ringer can be changed on the Grandstream phones? Mine sounds like the ringing you hear in a phone, not like the traditional ringing sound. Todd Wallace University of Phoenix Online Faculty [EMAIL PROTECTED] ___ Asterisk-Users

[Asterisk-Users] multiple simultanous calls with iconnecthere/delta three

has anyone configured asterisk to talk to iconnecthere (or any carrier) and have more than one line in/out. I want to be able to allow for multiple simultaneous calls. If so what product did you buy if not, is there any other carrier that can do this?? Todd Wallace

[Asterisk-Users] Company

Is there a way to allow someone to hit # for a company directory and step through the extensions? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Redhat driver for the X100P

I bought a X100P generic card that claims it uses the same wcfxo driver as the actual X100Ps. Where can I get this? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] recording calls

Is there a way or an "Open Source" product that allows youto record and/or monitor calls in progress? Todd Wallace

Re: [Asterisk-Users] A little bit of success

What kind of machine are you running on? Todd Wallace - Original Message - From: Nicholas Romero [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 3:15 PM Subject: [Asterisk-Users] A little bit of success Someone a little while back asked for comments

[Asterisk-Users] looking for a place to buy SNOM or Cisco Phones (Cheap)

Does anyone know where I can buy SNOM or Cisco (new or used) phones the cheapest. I need a few Todd Wallace

Re: [Asterisk-Users] looking for a place to buy SNOM or Cisco Phones (Cheap)

I have been watching for SNOM phones on ebay and have not seen any. There are plenty of Cisco phones, so I can definitely price those there. Todd Wallace - Original Message - From: Andrew Gillham [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 28, 2003 8:39 PM

[Asterisk-Users] iconnecthere

Has anyone made * to work with iconnnecthere's demo account? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] anyone with a used analog card for sale or trade?

Does anyone have a used analog card forsale or trade? Would prefer a 4 port, but beggars can't be choosey... Todd Wallace

Re: [Asterisk-Users] asterisk config files

I don't see that registry entry in windows 2000. is that an XP entry or should it be there in win 2000. I get it to register, but unable to make a call. Todd Wallace - Original Message - From: Anthony Wood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 24, 2003 8:59 AM

[Asterisk-Users] New here...

I am trying to get an initial setup up and going which I assume is a very common question here. My basic questionsare the following: Can I get Asterisk up and going without voice cards using it with SoftPhones internally as a proof of concept. (just calling extensions and leaving voice