similar to: Scrolling messages

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one phone setup as the receptionist phone, using hints to show busy office lines. This all works as expected. This is a new installation, and people are just starting to setup their phones. For those of you not familiar with SNOM phones, there is a row of keys on the right side of the phone which SNOM calls function keys. In

Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be

Hello All I'm settup my asterisk as belows: sangoma card, connected with E1, CAS Signalling. I have two problem. 1. The asterisk don't received any DTMF when caller input to 2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error. Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', but no exception

Hello All i have big problem for unicall. my system work successful with sangoma card, E1 and CAS signalling (vietnam). when at the some time. i have trouble then my system is half (CPU instructions = 100) i tested for some case as belows: - When i dial, then my system became answer, the caller hangup. system error message show (loop without condition and half machine) Jun 11 12:15:45

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten =>

Hi When I make a call from the outside to asterisk and the call is asnwered, all is OK, but when I make a call from the outside to asterisk and hangup before the call is answered, you got this WARNING in the console: Jul 6 19:33:08 WARNING[10037]: channel.c:1521 ast_read: Exception flag set on 'UniCall/2-1', but no exception handler Jul 6 19:33:08 WARNING[10037]: channel.c:1521

This is a serious problem! I have brought up this issue in four previous attempts to get some feedback. I find it hard to believe that no one else is having this same problem. Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible voice frame on Local/103@sip-00f3,2 of format alaw since our native format has changed to slin Apr 11 13:27:36 NOTICE[4446]: channel.c:1906

I am up to post 5 on my step-by-step but I hit a bit of a snag and so far my searches have failed me, I hope someone can help. (By the way, I added an asterisk index for quick navigation on the blog http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html.) Here is the snag and I am hoping for a little help from the collective. Inbound I have 2 different numbers. I can call in on both

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2

Hi, When I call forward on PAP2, the incoming call will right the forwarded number. However, there is one-way voice problem. The caller can hear the destination(the forwarded number), but after the called party answers, the caller can't hear anything. Then the CLI> produce continuous errors as following: Oct 6 10:57:45 NOTICE[11026]: channel.c:1409 ast_read: Dropping incompatible vo ice

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at