similar to: RFC2833 SIP trunks and DTMF

Hi, I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called "kpml". I wonder if Asterisk can support it. I found an

Hi, How do I listen to .sln audio file or convert it to some format that can listened to? Sox does not seems to support .sln as an input file. Thanks. - Andrew

Please can you provided me by a radius module name for asterisk,or how to authorize user and get cdr from radius server. Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an

Hi, I've managed to get X-Lite (v2 build 1050) working pretty well with *, but am having problems with the DTMF signalling. I've used inband signalling with no problems on the uncompressed codecs (G711), but obviously this doesn't work with the compressed ones (GSM). However when I try to use RFC 2833 it doesn't seem to pick up "0" properly. For example if I dial the

Just attended the Cisco Networkers conference and sat through a presentation addressing in part the Cisco IP phones. All was presented openly without the need for disclosures, etc. The following is simply an FYI from the presentation, FWIW. Existing IP Phones (7905/7912/7940/7960): Transport: UDP Call Signalling: rfc2833 Security: Digest Auth Feature parity to SCCP: None Sip Enhanced

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

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

Hello, I have installed oh323 channel driver. Outgoing calls to H.323 world do not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that userInputMode=RFC2833 has already been set. Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel? Kind regards, Fernando Herrera _____ De: Fernando Herrera [mailto:fherrera@iplan.com.ar] Enviado el:

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Hello, I'm having difficulty with registering a SIP account in a Snom 320 IP-phone. This is what sip debug tells me : [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] <--- SIP read from UDP:public_ip:58697 ---> REGISTER sip:sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;rport From: <sip:test3 at

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi, What are the current best practices for running asterisk as SIP B2BUA? Are there any sample configs online or the books that detail this configuration for the newbies? I'm going to run it behind 1:1 NAT for the clients in the public internet so I will use the externip, localnet, and nat settings. Thanks, Andrew

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the

On 05/12/14 16:46, Olli Heiskanen wrote: > INVITE that Asterisk (at port 5070) receives: > PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 > INVITE sip:660 at testers.com > <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP >