similar to: Question on AMI

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

I am getting this error from call manager (4.0) and asterisk 1.2.4 I have canreinvite=yes on the call manager setup. I can call into the asterisk box from call manager. THat seems to work. When I am calling out of the box using a call file I see this entry from call manager... What might be the problem with my setup? THanks, JErry ---------------- <Date>03/06/2006

I have the following dialplan. Everything seems good except for one thing. If the background message is playing and the user hangs up and does not press a digit how do I run an agi on that event. I tried an exten => h,1,agi(smvoice,-digium_failed) but that was never called. I am using 1.4.10 thanks, Jerry --------------------------- [smvoice-analog] exten => s,1,Wait(1) exten =>

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

Hi Asterisk Developers/users, I am facing a problem while using the cdr feature of asterisk(version asterisk1.4.24.1). Whenever I make a call using a ?*.call? file and it gets failed , it don't produce the CDR for that channel as it falls into ?OutgoingSpoolFailed? channel As there is no such channel defined for ?OutgoingSpoolfailed?.. I am using this line in extension.conf for capturing the

Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]),

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by

Hi all, I have setup my Cisco 79XX phone. Did the tftp, put the config files in the right location with the right names. Booted my phone, it does the tftp things, the screen shows my extensions everything seems fine. However, when I come offhook and try to dial 11 which is just a playback of demo-congrats in the dialplan the phone says Calling Out (INV) below is my sip.conf file - I presume it

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>

I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in

Some progress to report: I had to run asterisk as the user logged in - actually not even that. I could not "su user -c " to that user - I had to run it as that user. Then I did a test and got audio! Great... But when I do a second test. Asterisk HANGS on ChanIsAvail() Then I thought lets SKIP that - and just let it do the Dial() - I stopped everything - got it running again. - and

Hi all, I've got a program that creates a callfile and most if it working great. However, when a call fails, I'm trying to capture the reason, which I'm told should be in the ${REASON} channel variable. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Here is an excerpt from the callfile: Channel: local/155555555 Callerid:Tests <155555555> MaxRetries: 0 RetryTime:

I am running 1.4.22. I am doing a simple call into the dialplan and am getting a strange error: [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user "404" <sip:404 at 192.168.1.8>;tag=547521CB-DB0D6130 This is the only line that prints on the console... Typically I get a few lines like: -- Executing [33 at smvoice-sip:1]

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

hi All, I am using 11.25.2 and musing on hold. CentOS 7.4 I am trying to setup a MusicOnHold() streaming audio. I have one machine I tried this on and it worked. This machine is on the internet. I have another machine behind a firewall that does not play. Both machines installed mpg123. I have a windows machine behind the firewall that plays the audio stream so firewall is not the issue. I

hi I am getting this error: -- Executing [50 at smvoice-sip:1] Answer("SIP/440-0856dd70", "") in new stack -- Executing [50 at smvoice-sip:2] rtsp("SIP/440-0856dd70", "rtsp://192.168.1.175/img/video.sav") in new stack [Apr 30 11:22:48] WARNING[8031]: app_rtsp.c:1037 rtsp_play: >rtsp play [Apr 30 11:22:48] NOTICE[8031]: rtp.c:1287 ast_rtp_read:

-- Attempting call on DAHDI/1wwwwww for smvoice_callprogress at smvoice-dialout:1 (Retry 1) [Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request: Unknown option 'w' in '1wwwwww' [Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) Does DAHDI not know about the W ??? I think zaptel used

Hello: I have found the following problems with outgoing calls with asterisk, compiled with an updated CVS on 22 Oct. 1.- Problem with retries: Whenever I set the MaxRetries parameter, to something greater than 0 in a call-fille, Asterisk ignores the RetryTime parameter and retries every file in the outgoing folder when a new call-file is copied into that folder. So, if I make a call placing

I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working.... [smvoice-sip] exten