similar to: Unable to forward voice or dtmf

Hi all, This issue is giving me a lot of grief with my customers. I have 5 asterisk servers running in production, each one with almost 70 simultaneous calls at peak hour. Most of my customers complain that their calls drop after 20 seconds or so. After running through my cdr's, I see that the number of 20 second calls is MUCH larger than any other number. (see below) billsec count(*) 1 924

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

I'm trying to get my asterisk server to reinvite. I have two asterisk servers with public IP's. My users (behind NAT) register on one server (I'll call it server 1), and for some calls they are transfered over to the other server (server 2), because that server has the E1's. I want server 1 to be in the signaling path for billing reasons, but handling the media stream is killing

Hi, We've exhausted our internal capabilities as well as Sangoma tech support and were hoping someone with some expertise could help us with a pointer. Briefly, our issue is as follows. Periodically (several times an hour), we get either of the following error messages in our asterisk messages log. These correspond with dropped outbound calls on a one-to-one basis when the second error

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or

When trying to make outbound calls I am getting the Warning: File app_dial.c line 313 (wait_for_answer) Unable to forward voice. When making the call it attempts to dial (pounds are actually numbers but replaced to not show numbers we are dialing): Executing Dial("Sip/donas-bd7b", Zap/g1/1##########") in new stack Called g1/1########## Channel 1, span 1 got hangup **Above

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:

Hi Asterisk-ians! Need all of your help. I am stuck with this issue for last few days. I have one X100P installed in my system. My Asterisk is registered with another Asterisk Server/SIP provider as client and the call is successfully received by my Asterisk server (named as starwars). Now, the extentions.conf file states, the incoming INTO * should go out to fxo as below: exten =>

Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says "login incorrect". The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

Much to my surprise I tried to debug an AGI script today with "agi debug" on the Asterisk CLI and it did not work. Plus, I could find no reference on lie of it being removed. Is there another name for that command? I scanned the CLI help but found nothing similar. Both my 1.6 boxes do not have the command but my 1.4 box does. Thanks! Alex

Dear all I have recently install TE120P Digium E1 card now everything is fine and working i have connect my asterisk with avaya but when anybody transfer call from avaya i got this error on my asterisk consol [Aug 27 14:46:50] WARNING[19527]: app_dial.c:741 wait_for_answer: Unable to forward voice or dtmf -- Hungup 'Zap/32-1' I m waiting for your reply Satish Patel

Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b", "SIP/00919885497796 at sonetel")

Hi, I am trying to configure asterisk to translate between rfc2833 and inband DTMF. I have a cisco gateway which is configured as a trunk, and a cisco IP phone which is registered to asterisk. The gateway does not support rfc2833 and the IP phone does. I tried changing directrtpsetup to "no", and that didn't help. I tried changing "canreinvite" to "no", but that

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel: >-- Executing Dial("Zap/1-1", "SIP/6137451576@6477235412||t|") in new stack >Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such host: 6477235412 >Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create >channel of type

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

20 second timeout. If the call does not bridge or complete in 20 seconds, Asterisk considers it completed as failed. If you change the Dial to 30 seconds, the problem will become a 30 second one, etc... -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Doug Sent: Monday, April 19, 2010 9:12 PM To:

Hello mailing list, I have been porting one of my Asterisk boxes to 1.4 and I have encountered a nasty DTMF problem. What happens is someone might come in to my IVR and enter "12345" and what will actually come through could be along the lines of "12234445". Sometimes it works, sometimes it doesn't. I had this problem with 1.2 back in November but was able to solve it