[Asterisk-Users] IAX Phone Issues/McAfee Virus Scan vs. IAX Phone
--Request For Bug Reports-- I'm working on the next release of IAX Phone. Please let me know what, if any, issues you who use it may have run into. I hope to be able to release a new version in the next two weeks. Some fixes/features: - Conferencing - Proper handling of 'qualify' - Intercom - Paging - Phone Book --Virus Scanner Problems?-- I have been working through a number of the bugs already submitted to me. One is a rather large delay (300 ms+) for incoming audio. Some of this can be linked to the way it interfaces with the windows audio system. More of it, however, appears to be linked to Virus Scanning software. I tested on a system running McAfee 8.0. and found that the scanner software introduces the majority of the delay and causes other problems as well. The thread that runs the audio interface runs at high priority, but from time-to-time the processor spikes up due to some action on the part of the virus scanner and the audio drops out. Has anybody else experienced such issues with IAX Phone and McAfee? How about other virus scanners? Thanks, 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] VON Update - Greetings from Infomercial Central
The Hype Is Back Ok. Everybody who has ever been to a trade show knows that the majority of what you hear is marketing hype. That's to be expected. But usually you expect to see that on the exhibit floor or in companies' hospitality rooms. Unfortunately, the VON show seems to have decided to extend that to the Keynote Addresses and Industry Perspective sessions. By and large it looks like I have paid 2500 USD to listen to a series of the world's longest AT&T infommercials. It's hard to think of AT&T as a driving force in disruptive technology. This is not all bad. It is good research into the strategies the big players are using. It looks like the big money players (AT&T, Nortel, Siemens, Cisco, etc.) are really trying to push into VoIP in a big way. The other big positive is the fact that people are actually, well, positive. The buzz and the money seem to be back in the VoIP market. I just hope that the Asterisk community can take advantage of the change. Asterisk Team Arrives Mark and Greg, plus Jeremy from NuFone and Don Witt from Cylogistics are here and manning the booth. A good number of people wandered by to see the display and to ask questions about Asterisk. Lots of interest! I actually have seen an IAXy, so I can attest to the fact that they actually exist. Rumors have it that they will be available in the near future. Similar rumors surround the FXO modules for the TDM4XX platform. Astricon News Several more people have expressed interest in joining the Astricon tomorrow (Weds) evening. If we get more than 18 people we will change our location, so please show up in front of the conference center at 6:00 if you want to join us. You can also call me on my cell if you have questions: 816.806.8844. More as it happens. Live on the scene at VON. 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
RE: [Asterisk-Users] Codec between * servers
If you are using IAX between the servers, you can simply add a codec line to the iax.conf file to provide the specific codecs you want to use in the order you want to use them. Here's a sample disallow=all allow=ulaw allow=alaw If you want GSM as a fall back, tack on: allow=gsm The situation for SIP is pretty much the same. Google: http://www.google.com/search?hl=en&ie=UTF-8&oe=UTF-8&q=site%3Alists.digium.c om+allow+disallow+codec Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Monday, March 29, 2004 4:51 PM To: Asterisk Subject: [Asterisk-Users] Codec between * servers I was wondering if there is any information on how to select which codec is best to use between two * server. The local IP phones use G.711 to connect to the local server. I usually find that using a low end codec like GSM between servers will degrade the voice a lot, same with iLBC. Does anyone have any recommedation? -- Carlos Chavez Computer Engineer, CCNA Corporativo Lacer S.A. de C.V. ___ 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] FreeBSD
Not currently. There is a bounty for the development of working Wildcard drivers for Free/Net/Open BSD. Care to write them? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Moran Sent: Monday, March 29, 2004 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FreeBSD Do any of the Wildcards work with FreeBSD?? ___ 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 @ home ?
Call your local independent computer retailer and find either a retired PIII box or a low end Celeron box (I buy them in single-unit quantities for $300 each). Order the X100P from Digium. Configure. Call. Repeat if necessary. What kind of ideas are you looking for? Thanks, Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shawn Sent: Monday, March 29, 2004 2:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk @ home ? Looking for ideas on asterisk @ home. 1 snom 200 sip, 1 telephone line zatel 100, and 3 users.
RE: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group
The rollout is tomorrow evening, Tuesday March 30 from 6:30 to 7:30PM at the Hilton in Santa Clara, just across the street from the Conference Center. It's being held in a ballroom (I don't have the name with me, I'll try to post it later). I hope to see you there. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Monday, March 29, 2004 2:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group Where and when is the rollout meeting? I'd love to attend. Thanks! Paul Paul Mahler [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol Sent: Monday, March 29, 2004 11:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group The VON show has started off with a number of interesting announcements. First among these is a big announcement from Pingtel that they have created a not-for-profit corporation called SIPFoundry. This new company includes Pingtel (which has recently open sourced their SIPExchange PBX), Vovida and somebody else. Martin indicated in his presentation that the key goal of the new group is to leverage the open source SIP implementations to prevent legacy vendors (read Nortel, Avaya, Siemens, etc.) from using the "Embrace and Extend" model to co-opt and proprietize SIP. Pingtel (which makes SIP hardware) wants to keep the SIP market open and interoperable. They have a web site (which I can't seem to reach from the wireless network here at the show) for the new company/project: http://www.sipfoundry.org I spoke with Martin _ who gave the Pingtel presentation and is an officer of both Pingtel and the new SipForge organization. He indicated he would like to speak with Mark regarding the possibility of integrating the Asterisk community with the SIP Forge community. He indicated that Asterisk was not initially brought into the discussion only due to limited time/resources (and to a lesser degree because Asterisk is not SIP-centric). Can somebody out there take a look at the SIP Forge site and let us all know what the crux of the organization is set to be? They are having an open roll-out meeting tomorrow evening which should spell out some of the goals of the organization and the partners. Above all Martin wanted me to understand that he did not view the new open source organization as a competitor to Asterisk. What do you think? More on-the-scene reports to come. Thanks, 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 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] VON Update - Pingtel Creates new SIP Open Source Group
The VON show has started off with a number of interesting announcements. First among these is a big announcement from Pingtel that they have created a not-for-profit corporation called SIPFoundry. This new company includes Pingtel (which has recently open sourced their SIPExchange PBX), Vovida and somebody else. Martin indicated in his presentation that the key goal of the new group is to leverage the open source SIP implementations to prevent legacy vendors (read Nortel, Avaya, Siemens, etc.) from using the "Embrace and Extend" model to co-opt and proprietize SIP. Pingtel (which makes SIP hardware) wants to keep the SIP market open and interoperable. They have a web site (which I can't seem to reach from the wireless network here at the show) for the new company/project: http://www.sipfoundry.org I spoke with Martin _ who gave the Pingtel presentation and is an officer of both Pingtel and the new SipForge organization. He indicated he would like to speak with Mark regarding the possibility of integrating the Asterisk community with the SIP Forge community. He indicated that Asterisk was not initially brought into the discussion only due to limited time/resources (and to a lesser degree because Asterisk is not SIP-centric). Can somebody out there take a look at the SIP Forge site and let us all know what the crux of the organization is set to be? They are having an open roll-out meeting tomorrow evening which should spell out some of the goals of the organization and the partners. Above all Martin wanted me to understand that he did not view the new open source organization as a competitor to Asterisk. What do you think? More on-the-scene reports to come. Thanks, 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
RE: [Asterisk-Users] newbie question; can * screen calls?
Or, you could apply my patch, that I've been upgrading on the asterisk bug site. Check http://bugs.digium.com/bug_view_page.php?bug_id=752 Steve (and anybody else who may know about this code), I have the code for your privacy enhancements compiled and installed. My one question is, how do I configure it? Looking at the code, it appears that it reads from the internal DB1 database. Is this correct? Also, how do I enable the privacy options in the Dial(IAX2/|???). Thanks, 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
RE: [Asterisk-Users] Transfer from Grandstream BT100?
I have 1.0.3.81. How do you execute the transfer? Thanks, Steven -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV) Sent: Monday, November 03, 2003 8:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Transfer from Grandstream BT100? what version of GS firmware are you running ? I call from PSTN to GS, GS does xfer to XTEN, hang up GS call continues if you aren't running 1.0.3.81 or newer, then upgrade :) john brown chagres On Mon, Nov 03, 2003 at 06:15:02PM -0600, Steven Sokol wrote: > Hi, > > Does anybody know how to properly execute a transfer (without using the > |Tt option) from a GS100? Scenario: > > 1. I call from X-PRO (ext 1100) to Grandstream (1101). > 2. Grandstream answers. Call is established. > 3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold. > Grandstream gets dial tone. > 4. Grandstream dials 1103 (the extension of another GS100). > 5. Grandstream hangs up. > 6. Call is torn down. BYE messages sent from GS to X-PRO. X-PRO sends > BYE in response. > > This is the method suggested by the GS100 manual. I have tried both > using the [TRANSFER] and the [FLASH] keys, to no avail. Same with > pressing [TRANSFER] or [FLASH] to complete the transfer (instead of > hanging up). > > I can transfer calls from X-PRO without any problem. > > Thanks! > > Steve > > > ___ > 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Calls Don't Properly Connect (Continue Ringing) After CVS Update
I just noticed that messages like this: WARNING[1142106560]: File chan_sip.c, Line 451 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 8338 (Response) And this: WARNING[1142106560]: File chan_sip.c, Line 451 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 53696 (Response) Show up each time I try to place a call from the GS to the x-Lite. Never saw these before the update. I hope this helps. If anybody wants, I can pull a trace of the SIP stuff as well. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol Sent: Tuesday, October 28, 2003 10:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Calls Don't Properly Connect (Continue Ringing) After CVS Update Hi, I just updated my image from CVS, compiled and reinstalled it. Now whenever I make calls from my Grandstream phone to my X-Lite soft-phone, the call does not complete correctly. Scenario: 1. I take the GS off hook and dial 1100 (the extension of the x-lite phone). 2. The x-lite phone rings properly. 3. The user at the x-lite site answers the call. 4. The GS phone continues to "ringback" and does not detect that the call is complete. 5. After about 10 seconds the GS plays busy and the x-lite detects hangup. 6.The x-lite goes back on hook. This scenario was working properly (the call completed as expected) prior to the CVS update. Oddly, calls from x-lite to the GS complete properly and without incident. The big difference is that there is a "precursor" script on the GS extension that answers and plays the use's name using the name file in the voicemail folder. THEN it uses Dial to send the call to the SIP device. I swear there was a thread about this last week but I can't find it for the life of me. Perhaps it was in the error log at Digium. Any thoughts? Thanks - Steve ___ 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] SIP Calls Don't Properly Connect (Continue Ringing) After CVS Update
Hi, I just updated my image from CVS, compiled and reinstalled it. Now whenever I make calls from my Grandstream phone to my X-Lite soft-phone, the call does not complete correctly. Scenario: 1. I take the GS off hook and dial 1100 (the extension of the x-lite phone). 2. The x-lite phone rings properly. 3. The user at the x-lite site answers the call. 4. The GS phone continues to "ringback" and does not detect that the call is complete. 5. After about 10 seconds the GS plays busy and the x-lite detects hangup. 6.The x-lite goes back on hook. This scenario was working properly (the call completed as expected) prior to the CVS update. Oddly, calls from x-lite to the GS complete properly and without incident. The big difference is that there is a "precursor" script on the GS extension that answers and plays the use's name using the name file in the voicemail folder. THEN it uses Dial to send the call to the SIP device. I swear there was a thread about this last week but I can't find it for the life of me. Perhaps it was in the error log at Digium. Any thoughts? Thanks - Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compiling gastman under Win32
Where is the current Java binary? For that matter, where is the source for the Java version? Cheers, Steven -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Victor Medrano Sent: Friday, October 24, 2003 10:34 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Compiling gastman under Win32 Download binary with java , works fine with 2000 + Xp regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol Sent: Friday, October 24, 2003 4:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Compiling gastman under Win32 Can anybody tell me what I need to have in order to make/compile gastman on Windows using VC++? Or do I want to download and install gcc for Windows? I have never worked with anything designed to be cross-platform compilable. Thanks, Steve ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling gastman under Win32
Can anybody tell me what I need to have in order to make/compile gastman on Windows using VC++? Or do I want to download and install gcc for Windows? I have never worked with anything designed to be cross-platform compilable. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CTI interface(s) for Asterisk? [REALLY LONG MESSAGE - SORRY]
Hi. I have just finished Googling the list archive for information on doing CTI with Asterisk. I never found a definitive "here's how it's done" answer, but I did see a number of postings indicating that it was at least a possibility. (For those of you who already know about CTI, please pardon the pedantic introductions to each section.) Here are my questions: [First Party CTI (User Control Of Their Phone)] First party CTI allows application software on the user's/agent's desktop to control the functions of their telephone, and to receive call-related information with each call. Most of the time it is done by establishing a connection between the phone and the desktop directly. Sometimes it is done by establishing a connection to the PBX and telling it that you are the desktop UA for station X. In either case it allows users to place calls from within their PIM, and to pop screens based on caller information (generally ANI/Caller*ID). Questions: 1. Is there any generally accepted way of doing First Party Call Control for SIP hard-phones? Do we configure a "pseudo-soft-phone" to be the initial recipient of the call (thus grabbing all data related to the call) then deflecting the call to the actual hard-phone URI? If so, how do we monitor the "real" agent/station status? In the dreadful old world of proprietary PBXs/Phones, the phone manufacturer sometimes provides a 1st Party CTI interface in the form of a serial or USB connection on the phone, and a CTI driver (TAPI, JTAPI, TSAPI, CCXML, etc.) that runs on the desktop. Since we don't necessarily want to go pestering Grandstream, Snom, Cisco, etc. for CTI interfaces, is there a GOOD, ROBUST, SCALABLE way of doing this ourselves? 2. How about 1st party zaptel (analogue) CTI? Since this is much more along the lines traditional PBX connections, is there a good simple way to add Out-Of-Band CTI for these phones? Perhaps some kind of CCXML i/o gateway built into the code in pbx.c? [Third Party CTI (Global Monitoring/3rd Party Call Control)] Third party differs from first party in that an application has access to more than a single user agent device. 3rd party control allows for enhanced call routing based on ANI/Caller*ID, DNIS/DID, digits collected by an IVR, and also based on call center staffing and load (and probably lots of other factors). It provides omniscience (a god's eye view of the PBX) and omnipotence (a god's ability to control anything going on within the PBX). In the rotten old world of Avaya (my background) we have several third party CTI models to pick from. The most robust is ASAI on the Definity (MultiVantage) platforms. It allows you to: A) Grab a snapshot of a device, queue, or other entity; B) request and receive status monitoring information (events) about an entity; and C) 3rd party domain control (force a station off hook, move calls between queues, move agents from group to group, log stations into and out of groups, silently bridge calls to recording devices, place predictive calls and direct them to queues, etc). The ASAI interface is established by adding a small in-skin computer that monitors the internal signaling channels and sends commands across those channels to the primary call processing application. This separate device is needed because A) the primary processor is underpowered and B) because the primary processor runs a real-time tuned kernel and C) because they are the phone company and so they can do what they like. This ASAI interface can be accessed directly using some very lightweight drivers from Avaya (that cost big bucks) or by using and ASAI-to-TSAPI bridge that converts to/from ECMA CSTA (the formal TSAPI spec). This bridge normally runs on yet another PC and again costs an arm and two legs to install, plus other body parts to license and use. Question: What would it take to implement a similar (but hopefully more robust and easier to implement) system for Asterisk? Again I would think something more modern (like CCXML) would be a "best case" implementation, but there are a lot of legacy applications out there that could be integrated with Asterisk w/o modification if we added TSAPI and/or JTAPI and/or TAPI. Has anyone had any conversations with Mark about the best way to add this kind of functionality without impairing the near-real-time nature of Asterisk? Should this be posted to the Asterisk-Dev list instead? Does anybody have any experience at this kind of work? Comments please. Sorry for the REALLY LONG post. Cheers, Steven ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is the X100P a WinModem?
> hell i just got a quote today for D240JCT-1T1 for $4500ish
Yup. That's the one with the fancy DSP on-board that supports echo
cancellation ("continuous speech processing") and is generally used for
speech recognition apps.
By the way -- has anyone in the * community looked at adding ASR to the
astounding repertoire of services/applications integrated? Having just
arrived from the evil windows world, I really don't know what's
available in terms of Linux voice processing software.
It seems Mark and the Zapata team have worked out a very good
software-based echo cancellation/VAD system. That could be readily
integrated into a really cool ASR interface. Unfortunately I lack the
raw brainpower to build an ASR engine. Anybody else tried?
Cheers
Steven
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RE: [Asterisk-Users] Is the X100P a WinModem?
Coming from the [evil] Dialogic world (where even the drivers cost money) the prices Digium is charging seem very reasonable. New single-span Dialogic T1 interfaces cost at least three times ($1225 USD was the best price I could find on the D/240PCI-T1) what the single span Digium card costs. THEN ADD THE PRICE OF THE DRIVERS. THEN ADD THE PRICE OF REALLY BAD TECHNICAL SUPPORT. THEN ADD THE PRICE OF PROPRIETARY APPLICATION SOFTWARE. It's not pretty. (However, it has fed me and my family pretty well-...) The Quad-span card is even more of a bargain. The Intel boys claim that they have a serious advantage because they do voice, tone (and sometimes fax) encoding/decoding on the board, thus saving the core CPU. Big deal. Add another CPU. It don't cost much. Certainly not as much as the Dialogic quad-span DM3 card with the requisite DSPs attached. What I would love to see from Digium (after the new Wildcard TDM400P with FXO support is released) would be a higher density analogue platform. Sure you can keep packing cards in, if you have a high-end server with a passive back plane. But wouldn't it be nice to be able to do a 8-FXO x 4-FXS on a single card! Kind of like the Daytona cards that Pika offers. I know that channel banks can do much the same thing, but that adds one more point of complexity and would most likely cost more ($750+ USD for the bank + $500 for T100P = $1250 USD) than a good integrated high density card. Just my .02. Side question: how long will it be before telcos start offering commercial SIP or MGCP or other native VoIP services (over DSL/cable copper, fiber, wireless, etc.) and legacy stuff like voice T1 and POTS go away? Cheers Steven ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Running Asterisk and NAT on the same box?
Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the * using its PUBLIC (outside) address. Does this sound reasonable? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival hangs up?
Strange. I have a simple extension set up to do some Festival testing. (Festival 1.4.3 /w Asterisk patch). My extension looks like: exten => 1239,1,Answer() exten => 1239,2,Festival(Welcome to the asterisk system!) exten => 1239,3,Wait,1 exten => 1239,4,Hangup Some times it work right (sounds crappy but works as expected). Other times it picks up and hangs up without saying anything. The festival trace looks normal, but the * trace shows something like: Warning! [1122] Rtp.c, Line 374 (ast_rtp_read) Resource temporarily unavailable. Also, the TTS playback is sometimes cut off or truncated. An RTP error does not always accompany the cut off. System is dual processor P-III 866 MHz w/ 512 RAM. RedHat9. Current CVS of Asterisk. Any suggestions would be great! Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival, Patch, Asterisk, etc.
Hi, NOTE: Sorry if this gets duplicated. I sent a copy prior to joining the list. Newbie here with a repetitive question. How do I make the Festival TTS system work w/ *? I have festival installed (it came w/ RedHat9) but * complains: WARNING[1226530096]: File app_festival.c, Line 327 (festival_exec): festival_client: connect to server failed I have tested festival from the command line and it seems to work fine. I guess I need to make one or more festival servers available to *. How is this done? If somebody can point me to a series of previous postings that spell this out, I will go to the wiki and create a new section on Festival + *. I promise. Really. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
