I have configure 2 sangoma card each with 8 PRI lines with dahdi 2.6
the problem is i can see all channels configured in dahdi_cfg 480 channels configured but when I see /dev/dahdi i can only see 240 channels.
what could be problem I am using it wanrouter and when I put PRI in new card i only got calls on new line that means one of the card is inactive at same time all the lines and alarms are okay only suspected thing is /dev/dahdi.
is there nany setting in linux or kernel level which need to be set for solve…
We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels.
Lately we've been getting a disconnected calls. Keeping the consoles running it doesn't seem to be the PRI initiating the hangups, as I'll when I see hangups intiiated on the backend / PRI side:
-- Span 2: Channel 0/21 got hangup request, cause 16
Instead, I'm seeing
In a machine I've got : CLI> pri set debug off No such command 'pri set debug off' (type 'core show help pri set' for other possible commands)
CLI> core show help pri pri intense debug span
Hi, I have a PSTN Asterisk box that's connected to other dialplan PBXes through IAX2.
Recently this box was upgraded to 1.4.44 with the latest DAHDI version. I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI will return ISDN code 34 (as its supposed to do). However, the issue is that subsequent calls into that PRI channel are immediately responded by a Code 44 (channel not available) even though there is no live call on the channel.
Has anyone else experienced this behavior? Its a pretty crippling behavior since all of our channels eventually become unresponsive until a…
Can anyone help me find a specification for the 23B + D Asterisk TDMoE
(packetized PRI) link?
I am having occasional packets arrive from the Asterisk Server that do not have the standard four leading characters in front of the MAC addresses, and am not sure what this implies. For now, I discard such packets.
Also, I'm getting HDLC abort messages from the Server that occur almost exclusively in heavy call set-up traffic, despite that the PRI-style messages in both directions appear to have correct indices, Call Reference Numbers, etc.
If you can answer the question or if you…
I am using asterisk 1.4.43. When I call over the PRI to a single phone and play my recorded message its heard just fine. When I call over the PRI to a single extension (the switch then takes 3
phones offhook in intercom mode)
and play my same recorded message the audio is dropping out and the whole message is not heard.
What might this be? I think its the other switch - but if we use that switch to call that same extension and speak the audio does not drop out at all....
What can I look at?
Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH"
Below are the logs:
== Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660@voipphones:2] Dial("SIP/4856-00000003", "dahdi/g1/97052660") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-00000003' status is 'CONGESTION'
[trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel => 1-23
i would like to know if anyone has done or having idea regarding PRI terminations in Asterisk.
i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available
now i am thinking of arranging 8port sangoma card in this pci slots so i can arrenge 10 card in that.
is it possible to run system like that ? is it good idea , can asterisk handle 2400 calls if machine size and RAM is good.
let me know ideas and suggestions.