similar to: Cisco 79XX phones not talking to asterisk

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

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 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

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

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

I have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

Hi All Has anyone managed to get the hint priority with Swissvoice IP10S phones working? I have 2 phones: a Snom 360, setup as the reception phone on extension 11, and a Swissvoice IP10S on extension 12. When calling each other (tested both ways) I can only ever see the Snom 360 in the Active State from 'show hints'. The Swissvoice stubbornly remains in the Idle State when on a call!

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

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP

Hi All, Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf [general]context = demo ; Default context for incoming callsbindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV

To: asterisk-users at lists.digium.com From: webaccounts173 at jgoettgens.de Date: Wed, 29 Jul 2015 16:11:31 +0200 Subject: Re: [asterisk-users] Windows Asterisk Help Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf

Hi, do you have NAT between Asterisk and agent phones? S pozdravem Tomáš Holý Hi Tomas Thanks for replying. Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud. A typical sip.conf phone configuration on the remote server for the site is [general] session-timers=refuse disallow=all allow=g729:20 allow=ulaw

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

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, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

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

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer