[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 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] 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: http://lists.digium.com/mailman/listinfo/asterisk-users
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? I have added g.729 and need g.723. No. ___ 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] 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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 me. Any ideas where it look? I feel it is in the OH323 config.. Todd ___ 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] H323 calls drop on connect
Right after the call setup was completed... Todd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan Sent: Tuesday, March 02, 2004 4:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] H323 calls drop on connect Hi, Todd Did you notice that when you made the calls, were the calls indicated as answered, in both cases? And if so, did the indication answered pop up when the calls were actually picked up and answered or right after the call setup was completed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 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 me. Any ideas where it look? I feel it is in the OH323 config.. Todd ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ 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] OH323 errors
Did that also cause quality issues? Todd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, February 18, 2004 3:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-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 hit a 9+number and H323 when you hit 7+number. SIP is very clean, but I get 9+a scrolling Wrong Pitch 1st subfr. when you go out H323. Before, it was fine as the quality was fine, but now the quality is degraded so bad that I don't know if I have some other problem or this scrolling message causing problem. I have tried to track it down, but can't find a resolution to find my problem. SIP and H323 goes to separate carriers, but they are both using a Nextone. I am really at odds since, I can't seem to find out how to resolve. Any thoughts Todd ___ 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] OH323 errors
Thank you..Thank you..Thank you. Worked great! Also cleared up some quality issues I was having. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, February 18, 2004 3:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-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 hit a 9+number and H323 when you hit 7+number. SIP is very clean, but I get 9+a scrolling Wrong Pitch 1st subfr. when you go out H323. Before, it was fine as the quality was fine, but now the quality is degraded so bad that I don't know if I have some other problem or this scrolling message causing problem. I have tried to track it down, but can't find a resolution to find my problem. SIP and H323 goes to separate carriers, but they are both using a Nextone. I am really at odds since, I can't seem to find out how to resolve. Any thoughts Todd ___ 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] 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] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 fine as the quality was fine, but now the quality is degraded so bad that I don't know if I have some other problem or this scrolling message causing problem. I have tried to track it down, but can't find a resolution to find my problem. SIP and H323 goes to separate carriers, but they are both using a Nextone. I am really at odds since, I can't seem to find out how to resolve. Any thoughts Todd ___ 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] 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 via H.323, I seem to loose my IP address and the Nextone blocks the call. Seems to work fine when I go from SIP phone to SIP provider, but fails when I go from SIP Phone to H.323 provider. All outbound. When I look at the CDR's on the Nextone, I see 0.0.0.0 as my IP address. Any ideas?? Todd ___ 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] * 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 ___ 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] Is it possible to push the media processing off to a gateway for processing?
Well, I like the features asterisk gives me such as voicemail and IVR, Prompts, etc. I would like to offer an IP Centrex like service, but don't believe that I can handle very large amounts of users on a box. The reason I believe this is that the box would be doing all the media processing/DSP work on the processor and would be bound by the speed and memory of the box as to how many simultaneous sessions it could manage. A gateway has DSP's which are designed to handle this processing. I know they are more expensive, but I could handle large amounts of call volume this way and still keep the features asterisk offers. Another question I also meant to ask was having the ability to read extensions from a database instead of a .conf file. I was curious if anyone has asterisk pulling configs from a database like mysql. Todd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Friday, January 23, 2004 9:05 PM To: [EMAIL PROTECTED] Subject: Re: [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 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 (hardphones or softphones) then Asterisk can simply re-direct those calls to the gateway. Of course, Asterisk _is_ a gateway, so unless you have specific reasons for doing so, it would make more sense to use Asterisk to tackle those jobs with generic, cheap processing horsepower rather than expensive, proprietary DSP's. If you're just getting Asterisk to handle call setups and teardowns, why not just use a real SIP proxy for that? Or do you not know enough about your question to understand why I would differentiate between the two? (not being nasty here, just wondering if I need to explain more) JT ___ 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] 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 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] 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] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Notice Messages??? What does it mean there isn't already in the wiki... this is really a FAQ! btw, that means that your device is using silence suppression. since * doesn't support that, it issue the NOTICE below. that's not harmful, but if you're annoyed by those msgs, just turn off silence suppression in your device. Jan 16 15:56:11 NOTICE[240654]: File rtp.c, Line 263 (process_rfc3389): RFC3389 matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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] 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 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] detect third party
I have an application that I want to be able to do with asterisk. The scenario is this: Caller 1 is a user on asterisk. He/she calls mom dialing her phone number from his phone. Mom flashes and calls a third person and then bridges Caller 1 and Caller 3 through her phone. I do not want 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 for a large scale Implementation, after much customization of course. But it looks like a nice base. I was wondering if people could contact me either on the list or off list with examples of large implementations. In the range of 5K-30K active users. Best Regards, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgetone phones
Anyone ever have the Ethernet port on a Budgetone phone quit working. For some reason, it stopped link'ing up and I can't get an address from DHCP or when I set a static address, it would ping. I have reset to factory defaults and nothing seems to work. Feels like the port died, but nothing 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-users
[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/asterisk-users
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 is not complely established, so you are getting ringback from them. (Just a theory, but it makes sense in my head.) - Andrew Thompson Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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/listinfo/asterisk-users
[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
My phone number is being hosted by a provider and brought inbound on a Cisco 5300. A Nextone softswitch is in the middle passing the inbound call to me as a SIP request to my * box. He shows he is sending me the DTMF's, but I am not picking them up and interpreting them. I have tried info, 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 on people who have had any success with Asterisk. Well, I am here to report that there is some good news. My most heavily used Asterisk system has now been continuously operational for 21 days. Knock on wood. The system on average day passes between 8,000 and 10,000 calls. A little simple math is putting me closer the quarter million call milestone and well on the way to 1 million calls processed. Two more heavily used systems are now planned for the future. Crossing this milestone give me a little more ability to sleep at night. In seeing it I want to thank and congratulate all the people contributing a tremendous amount of work to extend and improve Asterisk. Cheers! -Nicholas Romero [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Subject: Re: [Asterisk-Users] looking for a place to buy SNOM or Cisco Phones (Cheap) Todd Wallace wrote: Does anyone know where I can buy SNOM or Cisco (new or used) phones the cheapest. I need a few Todd Wallace Uh http://www.ebay.com/ -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Subject: Re: [Asterisk-Users] asterisk config files On Fri, Oct 24, 2003 at 08:32:46AM -0500, Todd Wallace wrote: Would anyone mind sending me a working set of config files for asterisk and their softphone settings? I am really looking for very basic setup stuff as I This is real basic I have the make sample from Oct 14. I added the following to sip.conf: [woody] type=friend insecure=yes username=woody secret=bogus host=dynamic defaultip=192.168.2.76 [pyro] type=friend insecure=yes username=pyro secret=bogus host=dynamic defaultip=192.168.2.243 and the following to extensions.conf: exten = 1976,1,Dial(SIP/woody,15,tr) exten = 1974,1,Dial(SIP/pyro,15,tr) and then set up windows messenger (tools..options..accounts..Communications Server Account..Advanced) to connect to the asterisk using UDP only with the sign-on names [EMAIL PROTECTED] and [EMAIL PROTECTED], usernames woody and pyro, and passwords bogus. Look through extensions.conf for numbers you can call from messenger. Oh, also you'd need to change your windows registry HKEY_CURRENT_USER..software..messenger service..corpPCphone from 0 to 1. The Make a phone call comes up under your I want to... menu at the bottom of your messenger window. just want to show that the system works to my management before they will allow me to spend the money on phones and a telco card. Server is a redhat 7.3 w/ 512 RAM and dual 550's. Asterisk tar ball has been laid down, configured, make, make install, and make samples done... email address: [EMAIL PROTECTED] Todd Wallace -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 mail) Is there a jump start config that would accomplish this? What is the recommended SoftPhone that is "Open Source" Thanks!
