similar to: redirected call failure

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

Hi! I've a problem with caller presentation, when I call a number my presentation is set to "Presentation permitted, user number passed network screening", this works for outgoing call but not for incoming, numbers are always hidden and network provided number is always screened. I don't understand why? Anyone has an idea? My zapata options: usecallerid=yes

Salve, I'm somewhat stuck on how to get DTMF working with my setup and googling didn't yield anything similar. My setup consists of one CAPI-capable board (AVM Fritz!DSL) connected to a BRI (T-ISDN), one HFC-S board running in NT-mode connected to an internal S0 bus with some ISDN devices (DECT stations, TA) and, of course, some ethernet interfaces. ISDN standard used is Euro-ISDN.

Le mar. 15 janv. 2019 à 03:13, Lars Kruse <lists at sumpfralle.de> a écrit : > Hello Julien, > > > Am Mon, 14 Jan 2019 22:15:47 +0100 > schrieb Julien dupont <marcelvierzon at gmail.com>: > > > > ** Test 1 ** > > On VPN_office I use 'tcpdump -npi any icmp and host 192.168.1.3' > > When pinging 192.168.1.1 from client 1, with no success, I

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

Hello, Quick Diagram: Telco-PRI -> Asterisk <- Norstar PRI -> Norstar PBX (DMS100) (TE405P) (DMS100) | | V Cisco 7960G (SIP) I'm trying to change the Origination Number on my outgoing PRI, and running into a weird problem. If I make a call from a SIP extension off asterisk using the following context:

Hi, I have this setup: E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones Can someone tell me what's wrong with this call initiating from an analog phone connected to Alcatel PBX? It dies with NOANSWER but all works if I call other destination numbers. Dialplan is a simple Dial(zap/g1/0984465691) statement. At the end you'll find also zapata.conf.

I'm testing an ivr implementation (first time) using: exten => 620,1,Wait,1 exten => 620,2,Answer exten => 620,3,DigitTimeout,5 exten => 620,4,ResponseTimeout,10 exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for" exten => 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later

Hello Julien, Am Mon, 14 Jan 2019 22:15:47 +0100 schrieb Julien dupont <marcelvierzon at gmail.com>: > ** Test 1 ** > On VPN_office I use 'tcpdump -npi any icmp and host 192.168.1.3' > When pinging 192.168.1.1 from client 1, with no success, I see no packet > passing. Sorry - the tcpdump command should end with "192.168.1.1" instead of

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

Hi, When we make an outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup after answer. But when we place a full number before dialing everything is ok. Any help appriciated!! Thanks here is info with debug: == Primary D-Channel on span 1 up -- Executing Dial("SIP/200-164c", "zap/g1/|100|tc") in new stack -- Making new call for cr 192 --

Starting playing with asterisk 1.6.1.0 i found the following problems: In the cdr_pgsql, the sql statement is wrong: 2009-06-25 12:17:01 COT LOG: statement: INSERT INTO cdr

Hello, all. Could you tell me how to set the type of number in the outgoing SETUP message sent over PRI trunk? I need to have: Called Number (len=18) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '318989263037666' ] But always have: Called Number (len=18) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan

When receiving multiple calling numbers via a PRI for a call setup, I cannot find the ability to select between either first or last. Is there a way to do this currently? If not, can anyone help me in getting this to work? Here is the dump of the PRI in intensive debugging. Please note that I want the first calling number, not the last. Any help would be much appreciated. Below the PRI trace

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Hello, Configuration: Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1 I am trying to sort out the problem: 1. Provider's switch sends "SETUP"; 2. Asterisk receives "SETUP", rings allocated extension but does not send "Setup acknowledge" (or any other messages) to switch; 3. After 4 seconds of waiting of *'s response switch sends

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

hi! I need to pass the CLI for my outgoing ISDN PRI call from * box. here's the ISDN protocol debug. Q.931 > Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) > Presentation: Presentation permitted, user number passed network screening (1) '123123' ] > Called Number (len=12) [ Ext: 1 TON: Unknown Number

We are still having problems getting a Wildcard to work with a German E1 (PMX) interface. When starting asterisk it shows all B-channels starting up successfully (although our carrier told us only the first B-channel starts, if any at all). Incoming calls are not being signaled at all. (They seem to be intercepted by the carrier's switch, as no B-channel is up) Outgoing calls sometimes work,

Hi folks. My client is wanting to use call forwarding configured on their phones (Linksys SPA942), with a PRI from their provider. When we configure call forwarding, we invariably get a "The number you have dialed is not in service" message from the providers. Examining the detailed dial plan debugging as well as the PRI debugging, the number is dialed correctly. The only difference