similar to: [Fwd: Re: what is the magic needed from upgrading from 1.4 to 1.6]

I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. Is there a particular parameter needed for 1.6 that 1.4 did not care about? If I drop back to 1.4 it starts working again. Thanks Jerry

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

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

1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all

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

On Fri, Mar 10, 2023 at 9:49 AM Jerry Geis <jerry.geis at gmail.com> wrote: > > > On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis <jerry.geis at gmail.com> wrote: > >> I have a SIP trunk - calls going out work fine. >> >> Trying to setup an incoming call with a DNIS >> >> When I dial the number - I see nothing on the CLI. >> The person says

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

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 =>

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

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

I have a box running asterisk 1.4.17 that had been working. it has 2 uniden phones connected on it. This was working and now the phones dont ring when calling each other. below is the sip debug. I cant see why the other phone does not ring? I also tried changing the canreinvite for no to yes but that made no difference after restarting. Very simple network. server, linksys router and 2 phones.

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]

I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically "core show channels concise" "sometimes" I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF ]' I'm not

I have two calls (626 and 542) coming into the same phone (524). SIP/524-000005b5!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!9!SIP/542-000005b4 SIP/542-000005b4!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!542!!3!9!SIP/524-000005b5 SIP/524-000005b3!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!40!SIP/526-000005b2

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

I am using asterisk 1.4.18 with a polycom phone. sip.conf has: [532] type=friend username=532 secret=XXX dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=532 qualify=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm canreinvite=no I call into the dialplan and try to play demo-congrats and I hear nothing. Firewall is disabled. Everything is on the 192.168.1.X network for this

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:

[Dec 23 17:58:49] ERROR[3091]: chan_dahdi.c:8413 dahdi_pri_error: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX I am seeing the above error on DAHDI 2.1.0, asterisk 1.4.22 and libpri 1.4.7 I am using a TE120P card. I am also getting this VERY frequently: -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'DAHDI/0-1' Versus a normal hangup: