similar to: not succeeding to hide callerid with outbound calls

We use asterisk 1.6 with DAHDI and a PRI ISDN 30 line in the Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [s at macro-transfer:25] NoOp("SIP/joostkuif-00000003", "gehe= im") in new stack -- Executing [s at macro-transfer:26] NoOp("SIP/joostkuif-00000003", "voor= de SET CALLERPRES() =3D

We use asterisk 1.6 with DAHDI and a PRI ISDN 30 line in the Netherlands. Can anyone help me sorting out this issue?? Thanks in advance! -- Executing [s at macro-transfer:25] NoOp("SIP/joostkuif-00000003", "gehe= im") in new stack -- Executing [s at macro-transfer:26] NoOp("SIP/joostkuif-00000003", "voor= de SET CALLERPRES() =3D

Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-00000002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-00000002 -- DAHDI/i1/6463279153-2 is busy -- Hungup

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: >> >>

Hi all, I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode. Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34) which are only operating in dial out analog mode to deliver fax messages. After a while of running fine (50-200 dial out connections) on some S0 spans the following message occurs

Hi all, We're trying to hook up our Asterisk config (Card: TE410P) with a Siemens EWSD switch. The link is ok on both ends (green), with no errors. The problem is when we try to make a call from our side (via call files), we get the pri/E1 error Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) Our Telecom partner (they checked with Siemens) mentioned that we need to

Hi I have a small ISDN PABX , (BRI) at home - Siemens Gigaset 4175 which has cordless DECT extensions. I have set up asterisk on FC3 with two HFC cards and I am using the latest bristuff. I am trying to use * between my ISDN line and my PABX. One card is in TE mode and can receive and make calls OK. I can make and receive external calls to a SIP phone. I cannot get the other card to talk to my

Hi all, I am running to the following problem, when using the below dialplan to receive fax, everything works perfect till this line exten => receive,n,ReceiveFAX(${FAXFILE}): and then the following line cannot be executed, it's like asterisk can't go back to dialplan and continue, the good news is when i check what is received in my fax folder i find that the file is a valid one (not

I have an asterisk with a TE110P configured as T1 which is behind a PSTN gateway. This gateway has an E1 to PSTN and a T1 to asterisk. This T1 is configured as Network and * as CPE. Every call I receive in E1 gateway is directly switched to asterisk using T1. Remember E1 is alaw. Both E1 and T1 have Natural Microsystems boards with a very simple software. When I call to E1 asterisk signalling

Hi All, I've a BT Versatility PBX that I want to connect to my asterisk 11.9.0 box via BRI port in NT mode but actually I wasn't able to get it working. I've another standalone PBX, it's a Panasonic model, which works fine connected to the same port. The BT connected to a UK BT BRI line works quite fine after roughly a minute once connected. The PBX can be seen at

I managed to get Asterisk working with my PBX using T1, now I am moving on to trying to make PRI work. I have my zaptel.conf and zapata.conf configured as follows: Zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf: [channels] transfer=yes immediate=yes callprogress=yes language=en context=default switchtype=national signalling=pri_net group=1

I've been working on an asterisk box at work for a few weeks now, things were finally starting to sail smoothly until I hit this head scratcher this morning. It's a rather intricate problem, so bear with me. Heres the scenario. What works: If I call from my sip phone -> sip phone everythings ok If I call from sip phone -> external pots number ok as well If I map one of our

Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up" repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: < Unnumbered frame: < SAPI:

Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and

Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run "pri intense debug" and make a call to the BRI span, I can see a

Hi, I hope someone can help me with this.... Asterisk 1.0.6 Zaptel 1.0.6 Libpri 1.0.6, 1 Digium E100P card installed Panasonic TDA200 firmware v2.0.6 E1 Card Firmware 1.0.2 System is located in Australia, so as technologies go, I believe it is twist on the euro standard for the E1 signalling. Here is the situation. The TDA E1 card is set in cross over mode and I am using a functional

I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the

Can anyone work out why my callerid doesn't work on my isdn4Linux with asterisk (or without asterisk for that matter)... This used to work fine, and I am quite confident that the telco is sending callerid information (because they always do on all ISDN lines standard, only extra cost on POTS lines). This is the information from dmesg, whether asterisk is running or not: isdn_net: Incoming

Hi, I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with octobri card from Beronet. I use bristuff and have following zaptel.conf... # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=<span num>,<timing>,<line build out (LBO)>,<framing>,<coding>[,yellow] # # The timing

I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk setup with outgoing calls not completing and requiring an Asterisk reset to 'unstick' span 1. Sorry this is a bit long but I'm completely out of my depth :-( This system has been in use for some while and I recently upgraded it to asterisk 1.4.24, zaptel 1.4.11 and libpri 1.4.9. I didn't change