similar to: Festival RPM?

Displaying 20 results from an estimated 5000 matches similar to: "Festival RPM?"

2006 May 31
5
Converting .wav to .WAV
Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani
2006 May 24
5
macro-dial
Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani
2006 Apr 21
2
Asterisk on Red Hat AS 4?
Hi, I'm planning to install a new Asterisk server with a Digium TE410P card. Can I use Red Hat Advanced Server 4 (latest update)? Is this a good choice? Is recompiling Asterisk simple with kernel 2.6? Thanks -- Domenico Viggiani
2006 Jun 20
2
Snom 360 doesn't register after reboot
Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click "Re-register" in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani
2006 May 26
4
End of migration: adding support for some analog phones
Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I
2006 Jun 14
1
SIP call disconnected after answer
Hi, calling a partner on the other side of a SIP trunk, call gets disconnected immediately after answer. This is the content of log file: Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel: SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging channels SIP/232-2e41 and SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Hanging up
2006 Apr 28
3
Problems if GXP-2000 phones and Asterisk are not on the same network
Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? -- Domenico Viggiani
2006 Mar 31
4
IAX: Auto-congesting call due to slow response
Hi, I have a IAX2 trunk between two sites (connected with an high bandwidth link) but sometime/often I get: chan_iax2.c: Auto-congesting call due to slow response and call is dropped (and routed on a PSTN link). In iax.conf, I have: [iax-out] username=iax-in type=peer trunk=yes secret=xxxxxxxxxxx qualify=yes host=xxx.yyy.zzz.32 auth=md5 Any idea? Perpaphs is due to
2006 May 17
5
Plan to free myself from AAH
Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to
2006 Apr 10
1
"chan_iax2.c: Ooh, voice format changed to ..."
Can someone explain me this message: "chan_iax2.c: Ooh, voice format changed to ..." Where can I find a list of numeric codes used to identify voice format? Then, sometime I get an infinite loop of messages like these: DEBUG[15015] chan_iax2.c: Ooh, voice format changed to 1 WARNING[15015] channel.c: Unable to find a codec translation path from g723 to alaw DEBUG[15015]
2006 Apr 28
2
caching of sip account
Hi, during tests, I configured different SIP accounts on the same phone. Now I see this 'sip show peers output': Name/username Host Dyn Nat ACL Port Status 259/259 10.97.1.19 D 5060 OK (8 ms) 232/232 10.97.1.19 D 5060 OK (7 ms) where both extensions are registered and have the same IP. But now I have only one extension
2006 Jan 12
1
Problem with an automatic responder
Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR
2000 Mar 17
3
Bug in SMBCLIENT
I already posted this message but I had no answer. Sincerely, I think it is a bug and I'd like to hear developers on this. Platform: - HP-UX 11.00 - HP C/ANSI C Compiler (B.11.01.06) Copying a (large) directory structure from a NT share, interactive command: # smbclient //machine/share password >prompt >recurse >mget * fails to copy 76th, 115th, 154th file of
2006 Feb 03
4
CallerID popup
Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus
2006 Jun 14
2
Sangoma driver and zaptel
Hi, using Sangoma drivers: - doing 'lsmod', I see: zaptel ... wanpipe,wctdm24xxp,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 I'd like to avoid loading all these modules. What have I to do? - do I need to have 'zaptel' startup script under /etc/init.d ? Thanks -- Domenico Viggiani
2006 Jun 06
10
GXP-2000
I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I
2006 May 09
1
Shared call recordings with ARI!
Hi, I have '*1' in my features.conf file and I'm facing with a serious problem: - A and B are engaged in a call - C and D are engaged in a different call and decide to record their conversation hitting *1 - at the end, A and B are able to see C/D call recording using ARI with their user/pwd!!! Where is the problem? Asterisk or ARI? Thanks in advance -- Domenico Viggiani
2006 Jun 03
1
MWI lost after migration
Hi, I just migrated my Asterisk installation from 1.2.1 to another server with 1.2.8. Among a lot of things, I copied the whole content of /var/spool/asterisk/voicemail/default directory. All is OK but now I'm not able to see MWI indication for new messages on all my Grandstream GXP2000 phones (before migration, it worked). Peraphs do I missed something? Thanks -- Domenico Viggiani
2006 Jan 12
2
interfacing w/ a legacy InterTel PBX
Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for
2006 Mar 21
3
Zap<-->IAX codec?
Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --