similar to: low-level dump for PRI dchan debugging

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that

Hi all, My problem boils down to these errors: ... Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time This is triggered by lines in extentions.conf such as: exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W) The system is CentOS v5.2 with Asterisk 1.4.23 (druid-asterisk-1.4.23.1-2), a Sangoma A104

We are in the process of installing a PRI line and we are going to connect it to an Asterisk Box. Verizon called us today to find out some information. I am surprised that they have never heard of Asterisk or Digium. But anyways, they needed some information in order to set up the circuit. Does the TE110P support NI1 or NI2? (I think the answer is both) What is the number of digits

Greetings, This may be a bit arcane but does anyone know what the contents of a facility message should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on a DMS100 for the last four months to no avail. The message I am currently sending makes it to the switch but is returned with unknown message. Perhaps someone here has done it before and can help me out.

Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation.

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten => 12345678,1,Answer() exten => 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -----Original Message----- From: Erick Perez [mailto:eaperezh at gmail.com] Sent: Thursday, July 26, 2007 7:03 AM To:

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true?

We have install a DS3 with 28 DS1's we have an Adtran MUX breaking out the DS1's, we are trying to setup the system with 2 dchannels for each 4 DS1's. Everything looks fine when modprobe zaptel and wct4xxp and ztcfg -vvvvvv but when I asterisk asterisk it says: Aug 10 16:33:32 ERROR[8954]: chan_zap.c:6750 mkintf: Channel 24 is reserved for D-channel. Aug 10 16:33:32 ERROR[8954]:

Hi guys, Currently I just purchased the TE205P for the installation of my ISDN30 setup. I compiled, setup the card correctly I think. When I do the ztcfg -vvv it shows all 31 channels clear, with 30 bchannels using for voice ranging from 1-15, 17-31, and 16 is the dchannel. However, when I do a cat /proc/zaptel/1 I see only 24 channels configured, and then, when I do a cat /proc/zaptel/2 I see

Hello, I have following problem with Sangoma A104 card: CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI

In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a

Hello, We currently have a scenario where a large percentage of inbound calls on a telco PRI are intended for professors who aren't currently in their offices/at their desks. My question is, is anyone actively using the Asterisk "RLT" (Release Link Trunking) feature to bounce these sorts of calls back to the telco? The idea being to forward the call to their mobile phone without

Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27

Hello, About every 2 or 3 days I notice in the messages log file: Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 2: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 3: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 4: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 5: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on

I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: "very bad phasing reverb & feedback (from my rock & roll days)". This is quite intermittent, as in most cases, the user

I'm trying to get "blind transfer" from an incoming DAHDI line to an external number to work on an * 1.6 install using a T1 from XO. The documentation is very "distributed" and incomplete, so while it's not working, it's definitely more likely my error somehow. Couple questions if anybody is out there who even knows what TBCT is. 1) Is this even

Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31