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...tion 2111 is BUSY? Here are the debug logs: *CLI> *CLI> sip debug SIP Debugging Enabled *CLI> -- Starting simple switch on 'Zap/1-1' DEBUG[20491]: File chan_zap.c, Line 895 (zt_enable_ec): Enabled echo cancellation on channel 1 == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1", "SIP/2111 at gw1.langley") in new stack Interface is eth0 IP Address is 172.16.17.7 We're at 172.16.17.7 po...

...an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already tested: Xpro Logged in on Machine B using ILBC sending to Machine C and it works fine. Do send me your charges. Thank You, Rehan Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today!

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

I have servers that are running: Red Hat Enterprise Linux ES release 3 (Taroon Update 5) and Red Hat Linux Advanced Server release 2.1AS (Pensacola) [which I think RH renamed as Red Hat Enterprise Linux AS release 2.1]. The servers are running fine. Is it possible (and if so, how) to convert these to the equivalent releases of CentOS, without doing a complete upgrade (I want to mess with the systems as little as possible)? I work at a small c...

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?

I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1", "SIP/18504844535 at 18504844535)") in new stack WARNING[12299]: File chan_sip.c, Line 411 (create_addr): No suc...

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Im having problems setting up a trunk between two locations. Heres the setup I have: Server A is connected to the PSTN at my datacenter Server B is connected to a clients e1 line at his datacenter I only want to route calls from Server B to Server A and out through the PSTN. Server A has a lot of other things connecting to it, so I need a very specific context for all calls to go through.

Hello, I was wondering if it is possible to create a live data feed from Yahoo Finance stock data into an R program? Do any such modules already exist? Thanks for any help. Szymon [[alternative HTML version deleted]]

...Models, Biometrika, Vol. 65, No. 2 (August, 1978), pp. 297-303. One can still compute the correct p-value with >1-pchisq(value,correctdf) Nestor (R 2.2.1 on Linux, Suse 9.3) -- Nestor M. Arguea, Chair Department of Marketing and Economics University of West Florida 11000 University Parkway Pensacola, FL 32514 Phone: (850)474-3071 Fax: (850)474-3069

Please forgive me my ignorance ... I've spent two days trying to find out something about the format of the default configuration file, which CCM produces. The only example I have so far is the one from the chan_sccp source. There were tons of references on entering the callmanager commands on a cisco command line - which I don't have (don't need thanks to chan_skinny + chan_sccp).

...ect. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the default screen when I'm done making a call. Does anyone know how to fix this? --Eric -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free)

...cm but it didn't seem to make any difference. (set! default_after_synth_hooks (list (lambda (utt) (utt.wave.rescale utt .5 t)))) Does anyone have any suggestions? -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) -------------- next part -------------- A non-text attachment was scrubbed... Name: sanitizer.log Type: text/sanitizer-log Size: 1350 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/2...

...re it will have some bugs. It requires a few modules from CPAN and the asterisk-perl AGI interface. It's a very small script. Available at http://www.fnords.org/~eric/asterisk/ --Eric -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) -------------- next part -------------- A non-text attachment was scrubbed... Name: sanitizer.log Type: text/sanitizer-log Size: 1350 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/2...

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

When dialling in and dialling my extension, when answered I get " Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete --

Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA

I''ve been at the Ruby Hoedown in Raleigh, and have had a fantastic two days. I finally decided to get off my butt and really try to organize a LowerAlabama.rb ... Soooooo...... http://groups.google.com/group/loweralrb If you''re in Baldwin or Mobile counties in AL, or Pensacola FL, or Eastern MS...go sign up! I''m sure I''m not the only one down here. My plan is still to meet at the Barnes and Noble in Spanish Fort, unless someone has a better idea. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscrib...

I am wondering if it's possible to use a bunch of cards in a PCI backplane instead of going out to the extensions with T1 and then and adapter. How are people connecting to large amounts of extensions?

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia