similar to: Extensions, devices and dialplan

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

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

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

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

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

Hi, is there a RHEL4 RPM for the Festival text-to-speech system? Thanks -- Domenico Viggiani

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

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

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]

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

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

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

(Try again from the proper email address) --Rob -----Original Message----- From: Rob Thomas Sent: Thursday, 22 June 2006 12:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] syntax error That's freePBX or AMP code that we've since fixed - The replacement line is exten =>

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

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

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

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

Hi, I'm using AMP and its dialparties.agi as most important script in system. I'd like to port configuration to more embedded system, where I don't have Perl available. So I'd like to implement dialparties.agi functionality as closest as possible with dialplan language..... Are there any existing dialplan scripts-examples that are close related to dialparties.agi

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