search for: dahdichan

...428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory he...

...TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ; Telephone attached to port 1 dahdichan => 2 ; Telephone attached to port 2 [pstn] context=incoming-pstn-line ; Incoming calls go to [incoming-pstn-line] in extensions.conf signalling=fxs_ks ; fxs_ks Use FXS signalling for an FXO channel faxdetect=incoming busyd...

..._X.,n(dialfax),Dial(SIP/pbxfax/${EXTEN},30,r) exten => _X.,n,Hangup exten => _X.,n(dialanika),Verbose(2,Outgoing using pbxanika) exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r) exten => _X.,n,Hangup And here my users.conf: [00493511111111] fullname = luca secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgr...

...xten => _X.,n,Hangup > exten => _X.,n(dialanika),Verbose(2,Outgoing using pbxanika) > exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r) > exten => _X.,n,Hangup > > And here my users.conf: > > [00493511111111] > fullname = luca > secret = MYSECRET > dahdichan = 1 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > host = dynamic > dtmfmode=rfc2833 > canreinvite=no > sendrpid=pai > type=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transpor...

...== Using SIP RTP CoS mark 5 [May 29 19:42:13] NOTICE[2526]: chan_sip.c:20163 handle_request_invite: Call from '00493511111111' to extension '00493513333333' rejected because extension not found. users.conf on Ubuntu-PBX: [00493511111111] fullname = 00493511111111 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgr...

For those who don't know, (as I just figured out by reading the sourcecode), that all settings for a particular "channels" must be placed before the channel => entry. Ie, Immediate=no Channel=>1-24 Immediate=yes Channel=>25-48 Immediate=no Channel=>49-72 1-24 will have immediate set to no, 25-48 yes, 49-72 no Maybe someday the config will be

...ne_template](!) usecallerid = yes hidecallerid = no callwaiting = no threewaycalling = yes transfer = yes echocancel = yes echotraining = yes immediate = no context = longdistance signalling = fxo_ks [test1](phone_template) callerid = "Test 1" <(111)222-3333> setvar=myvariable=test dahdichan = 1 I have tried every example I have been able to find but nothing appears in a DumpChan. Thank you. Chet Stevens -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121201/13f68435/attachment.htm>

I am really a beginner of PRI ISDN board, I am wondering if there is a "quick start" chan_dahdi.conf configuration I could use. I tried to install two "FreePBX" boxes follow the instructions from "http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html" connected them between PRIs, It worked. And now if I refer the FreePBX "chan_dahdi.conf" it looks

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...callreturn = yes echocancel = yes echocancelwhenbridged = yes relaxdtmf = yes rxgain = 0.0 txgain = 0.0 group = 1 callgroup = 1 pickupgroup = 1 immediate = no context = myphones signalling = fxo_ks [phone1](phone) signalling = fxs_ks callerid = "Andrew F Robinson" <(503)543-2338> dahdichan = 1 [phone2](phone) signalling = fxs_ks callerid = "Michael C Robinson" <(503)987-1322> dahdichan = 2 [phone3](phone) callerid = "2010" <2010> dahdichan = 3 [phone4](phone) callerid = "2011" <2011> dahdichan = 4 [root at robin asterisk]# extensio...

...ernet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180 Then I added the peer in my users.con: [00491771111111] fullname = 00491771111111 secret = MYVERYSECRET type=peer nat=yes qualify=yes qualifyfreq=60 hassip = yes dahdichan = 1 transport=udp,tcp callwaiting = no context = default host = dynamic dtmfmode=rfc2833 dial=SIP/00491771111111 and finally "core reload". On my Gateway I configured the NAT so: /sbin/iptables -t nat -A PREROUTING -p udp --sport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin...

Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC

...ext = DID_span_1 zapchan = 1-23 --- /etc/asterisk/users.conf (asterisk 1.4.24[0-1] w/ SVN-branch-2.0-r4661 GUI) --- group = 1 hasexten = no switchtype = national signalling = pri_cpe trunkname = Span 1 trunkstyle = digital ; GUI metadata hassip = no hasiax = no context = DID_span_1 zapchan = 1-23 dahdichan = 1-23 --- The dial command above did not work on the second group of servers. Instead I had to change the group to 1 thus:

...elease. Do the log notes of the CED tone or the T.30 ECM warning have anything to do with this? chan_dahdi.conf: [pstn] context=incoming-pstn-line ; Incoming calls go to [incoming-pstn-line] in extensions.conf faxdetect=incoming faxbuffers=>6,full ; who knows what this does busydetect=yes dahdichan => 4 Any idea on how to debug this? sean

...ernet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180 Then I added the peer in my users.con: [00491771111111] fullname = 00491771111111 secret = MYVERYSECRET type=peer nat=yes qualify=yes qualifyfreq=60 hassip = yes dahdichan = 1 transport=udp,tcp callwaiting = no context = default host = dynamic dtmfmode=rfc2833 dial=SIP/00491771111111 and finally "core reload". On my Gateway I configured the NAT so: /sbin/iptables -t nat -A PREROUTING -p udp --sport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin...

...UI metadata busydetect = yes busycount = 3 busypattern = 480,620 ringtimeout = 8000 answeronpolarityswitch = no hanguponpolarityswitch = no callprogress = no progzone = in usecallerid = yes cidstart = ring pulsedial = no cidsignalling = v23 flash = 750 rxflash = 1250 mailbox = callerid = asreceived dahdichan = 1 context = DID_trunk_1 group = 1 hasexten = no hasiax = no hassip = no registeriax = no registersip = no trunkstyle = analog disallow = all allow = all gui_volume = 2 ; GUI metadata signalling = fxs_ks gui_fxooffset = 0 ; GUI metadata rxgain = 0 txgain = 0.0 channel = 1 /dahdi/system.conf # S...

...is article: http://www.linuxjournal.com/article/9399): sip.conf: localnet=192.168.200.0/255.255.255.0 localnet=192.168.20.0/255.255.255.0 externhost=mypc.noip.com externrefresh=180 rtp.conf: rtpstart=10000 rtpend=10100 users.conf: [00491773333333] fullname = 00491773333333 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=yes qualify=yes qualifyfreq=60 avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00491773333333 A...

...yes rtp_symmetric=yes force_rport=yes [registration0](!) type=registration transport=0.0.0.0-udp retry_interval=60 max_retries=10 expiration=3600 auth_rejection_permanent=yes server_uri=sip:172.16.25.23 [fxs17](endpoint0) context=from-sip-fxs aors=fxs17 outbound_auth=fxs17 from_user=1121 set_var=DAHDICHAN=17 [fxs17] type=aor qualify_frequency=60 contact=sip:1121 at 172.16.25.23 [fxs17] type=auth auth_type=userpass password=111111 username=1121 [fxs17](registration0) outbound_auth=fxs17 client_uri=sip:1121 at 172.16.25.23 contact_user=fxs17

Hi, Built a new server at the weekend and install Asterisk 1.6.0.9 and IAX and SIP work great :) The one problem I am having is getting the OpenVox (TDM400 type card) to work. It is successfully identified using WCTDM kernel module and dahdi_scan picks it up just fine. The issue is when I try and setup dahdi_channel.conf as it fails everytime. When running asterisk -rvvvv I see the port pick

PRI debug of the entire call would be great, also, switchtype would be awesome as well. Thanks! Matthew Fredrickson On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas <crt.rojas at gmail.com> wrote: > Hi > > Did you activate the pri debug on the cli asterisk? > > On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez <cursor at telecomabmex.com> > wrote: >> >>