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...there is a smoking gun--I dialed "8" on the Budgetone (192.168.1.21) to get voicemail on the asterisk box (192.168.1.10). Thanks. B. -------------- next part -------------- Sip read: INVITE sip:8@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.21 From: "BC IP Phone" <sip:btel@192.168.1.10>;tag=c3cedeba-47c2-6790-8eb4-5b15010f6079 To: <sip:8@192.168.1.10> Contact: <sip:btel@192.168.1.21> Call-ID: 95eae8bb-9fc7-1f5d-bb7d-23eddebaf21c@192.168.1.21 CSeq: 39684 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOT...

...h codec pass-through. I have two SIP phones, both with g729, behind two Asterisk servers. I set all the configs, SIP and IAX, to "disallow=all; allow=g729" on both servers. But the originating server won't even try to call the destination server: -- Executing Dial("SIP/btel-c7d7", "IAX2/bris/10101") in new stack Mar 5 02:55:32 WARNING[2786]: channel.c:1942 ast_request: No translator path exists for channel type IAX2 (native 63508) to 256 Mar 5 02:55:32 NOTICE[2786]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'IAX2' (caus...

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

...ival version of the ping is option 32. Please report problems and/or issues directly to me. I'm trying to get ping.agi stable enough to release the source code for it under GPL. Thanks in advance, Eric aka ManxPower -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown

...to ast-1 and I dial 12010 I assume ast-1 asks ast-2 to resolve the extension 12010, and I also assume that ast-2 returns "exten => 12010,1,Dial(Zap/1)" then ast-1 tries to Dial(Zap/1) which is not an interface on ast-1 and the call fails. What information am I missing? --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Anyone know what model and what support the new $100 Cisco has? http://biz.yahoo.com/djus/030428/1030001060_1.html -- Steven Critchfield <critch@basesys.com>

And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw

...card X100P's which can > be obtained on eBay for $9-$20, usually. Much cheaper price-per-port, > although the TDM would give better expandibility. You mean NON Digium X100P's. Digium no longer sells the X100P. The cheap ones on eBay are "clone" cards. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

...asterisk/modules/chan_sip.so: undefined symbol: ast_pickup_ext WARNING[1024]: File loader.c, Line 319 (load_modules): Loading module chan_sip.so failed! I totally deleted my local asterisk source tree (asterisk, zaptel, zapata, and libpri) and it still happens. Anyone have any ideas? -Eric -- BTEL Consulting 504-595-3916x2111 (Experimental) 850-484-4535 (Office) 877-552-0838 (Cell)

...[2-9]#,2,Background(${MYSOUNDS}/enterspeednumber) exten => _X.#,1,SetVar(SPEEDNUMBER=${EXTEN}) exten => _X.#,2,Background(${MYSOUNDS}/speedcode) exten => _X.#,3,SayDigits(${SPEEDCODE}) exten => _X.#,4,Background(${MYSOUNDS}/assignedto) exten => _X.#,5,SayDigits(${SPEEDNUMBER}) -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

I'm getting the following message when I start Asterisk: WARNING[1024]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 Did I mess up the registration key or is something else wrong? --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

...on the iaxtel.com web site to see if my asterisk is registering and what 700 number is associated with my iaxtel account? I registered many months ago but never used it. My asterisk shows registered, but I can't seem to receive any calls (callers get a the user is not registered message) -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

...uot;Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten => 3009,4,VoiceMail(u${RDNIS}) exten => 3009,5,Hangup -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

Tie line is type em_w (old school stuff) to TE110P. Phone rings on the asterisk side, but the calling party does not hear the ring through sound. If I pick it up within the first two rings it goes through and I can talk otherwise our old switch drops the call. Anyhow...here is my config if anyone can shed some light on it. It used to work with HEAD a few weeks ago. -Matt

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

...vailable with Packet8 is at http://www.packet8.net/about/areacodes.asp I have not gotten Asterisk to work with Packet8, however I have put very little effort into making them work together. It's easier to just plug the DTA310 SIP/POTS device that you get from Packet8 into a X100P. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. thanks, Somesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/e26855c9/attachment.htm