similar to: Bad DTMF detection

I've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information

Hi list, Just installed spandsp. In my limiting testing, I have an issue on a Philips fax machine (HFC21) directly connected to my * server through TDM400, reception with rxfax works fine, but txfax always fails. Below is a transcript of failed transmit. This is with asterisk-1.0.3 (with native moh patch but I don't think it is the source of the problem). I already tried libtiff 3.5.7,

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

Anyone have a suggestion on where I can get a decent new MB with 5v capable PCI slots. It seems like every decent server MB on the market has 3.3V slots only. Is diving into the junkbin my only choice if I can't afford to replace the 5v quad-T1 wildcard? Thanks Mark Farver

All: I am trying to use Voicepulse as my incoming line and want the caller to simply dial the extension of the party they want to reach. Here is my problem: - the first time they dial it works fine and I see the following on my console Aug 24 23:14:31 DEBUG[-1126876240]: chan_sip.c:4408 build_route: build_route: Contact hop: <sip:6035057098@66.234.228.137> --

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

I have a pair of sites tied together with a T1 line running zaptel PPP on either end. The system works, but I keep getting these messages in the kernel logs, and users are reporting connection problems (TCP timeouts, and hangs) especially under high usage. --snip-- HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 Log for a call from mitel heading outbound: ------------------------- -- Accepting call from '' to

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:

Our setup is: 9.6k Modem <-analog-> Mitel SX-200 <-(pri)-> Asterisk <-(pri) -> Telco The modem works fine with the Mitel directly connected to the Telco, but once we add Asterisk in between connections start failing. I suspect the issue is caused by the echo canceller, since I believe the issue appear about the time we turned echo cancellation on (for the IAX users). We

After sometime I got my E1 PRI configured correctly in /etc/zaptel.conf and I now don't see any alarms on the E1, but I can't still dialout correctly, I enable every debug that I could though of and this is what I see: At the end the D-channel is down and I cannot even try to connect another call because it tell me that the zap channel is unavailable ,the other this that I notice is that I

Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten => s,1,Answer Exten => s,n,SayDigits(?1?) exten => s,n,Festival(hello john) exten => s,n,Hangup I use call files to

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Hi. I have the following line in the default context of all my internal extensions: exten => 9876,1,Transfer(125) When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want. Unfortunately when I dial 9786 from my Zap connected analogue phone, the transfer doesn't go through and the dialplan drops through to a hangup. debug

I just loaded from CVS this afternoon and in the debug output I see... DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: m on Zap/16-1 DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: u on Zap/16-1 I knew about DTMF 0-9, A-D, *, and #, but I didn't know about m and u :-).

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer