search for: channeltyp

Hello, I'm currently evaluating asterisk 13 (Currently on 11). We're testing the migration from SIP to PJSIP. Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150126/eeb8cc1e/atta...

...Let's drop the hangup handler at the moment, and focus on the "saving to file" part. Then my issue is I can't update CDR value is hangup exten. Here is a dialplan that illustrate this: [from-foobar] exten => _X.,1,Verbose(0,Entering context ${CONTEXT} from channel ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same => n, Set(CDR(userfield)=foo) same => n, Dial(SIP/foobar/${EXTEN}) same => n, Set(CDR(userfield)=bar) same => n, Hangup() exten => h,1,Verbose(0,Entering context ${CONTEXT} from ${CHANNEL(channeltype...

Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI> core show channeltypes I would have response like: connect*CLI> core show channeltypes Type Description Devicestate Indications Transfer ---------- ----------- ----------- ----------- -------- USTM UNISTIM Channel Driver...

*Good morning! * *I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the necessary configurations that should make! I have been installing the trixbox version 2.6.1.13 and a Digium 110p, they put in the trixbox only get to do to work in ISDN! Tha...

Is there any reason why IAXPeers output is different from SIPPeers output? The response has no Eventlist: start Ej. Response: Success Eventlist: start Message: Peer status list will follow Event: PeerEntry Channeltype: SIP ObjectName: 1001 ChanObjectType: peer IPaddress: 192.168.175.1 IPport: 63772 Dynamic: yes Natsupport: no VideoSupport: no TextSupport: no ACL: no Status: Unmonitored RealtimeDevice: yes Event: PeerlistComplete EventList: Complete ListItems: 1 Response: Success Message: Pe...

...reasonable output from ztcfg -vv, and zttool shows the installed module (TDM400) with one FXS module. But when I start asterisk, I get an error saying that my IAX connection won't work in trunked mode because there's no timing interface. Zaptel doesn't show up in the output of show channeltypes. Should there be a problem with using the trunk version of zaptel, but 1.2.16 of asterisk? Are there any places that I can specifically load/enable the zaptel module? Any help much appreciated before I go insane... J Regards, Mark. -------------- next part ------------...

I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know what would be the best Distro of Linux to use and version, what version of Asterisk works best to interact with CallManager, and what H323 ChannelType works. As you probably read in another thread I tried FC5 with Asterisk 1.4 and OOH323 (included with the addons package). This doesn't seem to work to well, as somewhere along the line either CCM or OOH323 is disconnecting the call as soon as the playback application is run. --------------...

...libpri 1.4.1 mISDN 1.1.5 & mISDN usertools zaptel 1.4.3 installation went ok, "mISDN config" recognized my card & created the accpording /etc/mISDN.conf file. I manually edited /etc/asterisk/misdn.conf. Asterisk is loading the chan_misdn and lists mISDN when issueing "show channeltypes" - however it indicates "Devicestate - No". when I look for "misdn show stacks", it lists the single port of the ISDN-card, however indicates "L2Link DOWN, L1LinkDOWN". so I guess theres something wrong, unfortunately I've no idea on what to check next......

Hi Guys! I got a Diva passive ISDN card and I can't get it work with asterisk 1.4, It is supported in the kernel as an isdn4linux device but I can't find "Modem" channel type when i type in: core show channeltypes. I'm guessing it is removed in asterisk 1.4. Tried with capi interface but it does not work :( Anybody got some idea how can i make it work or got a link to a working how-to? Thank you. pc:~# capiinfo capi not installed - No such device or address (6) pc:~# lspci -v 00:0b.0 Network con...

...onfigure the h323.conf? i want asterisk and gnugk in the same machine (lets say ip 192.168.0.10). then sip_client at 192.168.0.100 (telephone number assigned for instance 100) and H323_client at 192.168.0.200 (telephone number assigned 200) how do i configure the files to do this? if i type show channeltypes i see: Type Description Devicestate Indications Transfer ---------- ----------- ----------- ----------- -------- OOH323 Objective Systems H323 Channel Driver no yes no SIP Session Initiation Pro...

Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online,

.... I hve connected the wall-phone-input to the "line" slot and "phone" to my home-phone. I do not hear any dial-tone on my home-phone. Asterisk seems to recognize my hardware....here are the relevant logs/configs: Any help would be highly appreciated. jserver*CLI> core show channeltypes Type Description Devicestate Indications Transfer ---------- ----------- ----------- ----------- -------- Phone Standard Linux Telephony API Driver no yes no Local Local Proxy Channel Driver...

...st, I'm trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on asterisk-13.3.2, but they don't return anything. Is this a bug, or did I miss something? Here is my test dialplan: exten => *98,1,Answer same => n,NoOp(Channel=<${CHANNEL(name)}>,type= <${CHANNEL(channeltype)}>) same => n,NoOp(AOR=<${CHANNEL(aor)}>, contact=<${CHANNEL(contact)}>) same => n,Set(aor=${CHANNEL(name):$[LEN(CHANNEL(channeltype)) +1]:-9}) same => n,Set(contact=${PJSIP_AOR(${aor},contact)}) same => n,NoOp(URI=<${PJSIP_CONTACT(${contact},uri)}>) same => n,N...

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

...; > > In diaplan, how can you know the technology and channel of an incoming > > call ? > > I was thinking of something like : > > > > [incoming] > > exten => _XXXX,1,Set(CALLERID(num)=00${CALLERIDNUM}) > > exten => _XXXX,2,NoOp(This call comes from ${CHANNELTYPE}) > > exten => _XXXX,3,NoOp(This call comes from ${CHANNELNO}) > > exten => _XXXX,4,NoOp(Said differently this call comes from > > ${CHANNELTYPE}/${CHANNELNO}) > > > > > > My ultimate goal is to have this working with Zaptel channels (from a > > bris...

...the codec is already chosen before the call is made, and you > told it that g722 is permitted. > > There are all sorts of discussions in play about codec negotiation, > but at this point in time, if you want different behaviour you'll need > to go and code it yourself, and cross-channeltype this is not going to > be trivial :) > > Cheers, > Steve > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar ever...

...uot;) in new stack -- <SIP/403-097cc8e8> Playing '/var/lib/asterisk/VOD/jolin-512k' (language 'zh') cwhuang*CLI> <--- SIP read from 172.16.148.129:36042 ---> <-------------> --- (0 headers 1 lines) --- cwhuang*CLI> show ch channel channels channeltypes channeltype cwhuang*CLI> show channel SIP/403-097cc8e8 -- General -- Name: SIP/403-097cc8e8 Type: SIP UniqueID: 1190258125.131 Caller ID: 555 Caller ID Name: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormats:...

I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the provider I have deny=all and allow=ulaw. This can cause potential audio degrading and wastes cpu cycles.

..., the callee can open the door by pressing #1, and the dial plan for extension 1 takes care of the rest. This works when I make a typical SIP to SIP call, but it doesn't when I call from the console, using chan_alsa. I can see that the transfer feature is inactive: rasterisk*CLI> core show channeltype console -- Info about channel driver: Console -- Device State: no Indication: yes Transfer : no Capabilities: 0x40 (slin) Digit Begin: no Digit End: yes Send HTML : no Image Support: no Text Support: yes However, I am unable to find a way to activate it. How can I tr...

...------------------------ * ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport' When Running "sip show peers" (Reported by Michael L. Young) * ASTERISK-22659 - Make a new core and extra sounds release (Reported by Rusty Newton) * ASTERISK-22919 - core show channeltypes slicing (Reported by outtolunc) * ASTERISK-22918 - dahdi show channels slices PRI channel dnid on output (Reported by outtolunc) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0 Thank you f...