similar to: CallerID inconsistently presented through ISDN/cellular networks

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

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

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

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All

Hello List, it is a very long time since I wrote here.... It has been still in Zaptel times.... Today I am run into a related problem: I can't get a DAHDI setup to work 100%. I am configuring an Astribank XR00013 (BRI, two ISDN ports). At some degree the installation (latest DAHDI drivers, Asterisk 1.6.0.18 on Centos 5.x) works. I can get incoming calls to the dialplan context that is

Hi, After a long discussion with a friend, I would like to ask here: 1.According SIP RFCs, is possible/recommended to have different values in >From and P-Asserted-Id fields ? For instance, From field showing 123456789 and P-Asserted-Id showing 987654321 (beside privacy considerations) ? 2. When Bob forwards to Cory a call coming from Alice, would expect Diversion/History-Info header to

Perhaps I've got the whole concept wrong, but here goes: Using 1.4, when someone from the outside dials my direct line (123456), I want it to call my extension at work (SIP/456), my extension in my home office (vpn connection to corporate lan, SIP/678) and my mobile (654321). So my dialplan is thus: exten => 123456,1,Dial(SIP/456&SIP/678&Zap/G3c/07803654321,30) exten =>

Dear Asterisk-Users: Question: ======== How do I get asterisk to pass DNID/RDNIS information between asterisk machines using iax2, in a Dial(IAX2...) command ? Setup: ===== I have two asterisk boxes, MASTER and SLAVE. MASTER is running 1.2.0 and SLAVE is running 1.2.1. The main box handles incoming calls on a multiple lines (both via hardware connection to our internal PBX and calls

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

I have a PRI E1 link between Asterisk 1.4.24 and Alcatel-Lucent OmniPCX Enterprise R9.0. As EuroISDN it works fine. However, I need to move to QSIG because of a firmware upgrade on the Alcatel PBX which doesn't support EuroISDN (please don't ask why). Besides, I've read somewhere that 2 B Channel Transfers "should" work with * 1.4, the latest 1.4 libpri and QSIG. So this

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

Below is a snipped debug on our PRI. We are getting number only for the CallerID but the telco says they are sending us Name and Number. We are getting the Name in a second frame but Asterisk is not passing it to the device it rings. The message below says "Presenation allowed of network provided number" which leads me to believe Asterisk thinks it should not be displaying it. Can anyone

I there, I have some trouble to do working caller id name for outgoing calls on the PRI we just installed. Caller id name work on incoming calls only. Caller id number work on incoming and outgoing calls. The provider, Goup Telecom, said that's in what i'm sending. They said that I send the cid name in ascii code and to do it working, I need to send it in hex. So I take some traces

I have some Cisco 7905 phones with the SIP load 1.02.00(040406A). When the phone is off-hook but no call has been placed, or when the Do Not Disturb is activated, the phone returns a 302 "Moved Temporarily" message back to asterisk as follows: ----------- -- Executing Dial("SIP/5060-0811bb00", "SIP/9871234|20|Ttr") in new stack -- Called 9871234 -- Got SIP response

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

Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have two HFC-s boards I configured in NT and TE mode respectively. When I connect the two boards together, I can dial extensions and I see the correct called and caller ID numbers: -- Executing SetCallerID("Zap/2-1", "7516862") in new stack == CDR updated on Zap/2-1 -- Executing Dial("Zap/2-1", "Zap/g2/0795025602|30|r") in new stack --

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass

Hi All, In my dialplan and standard asterisk CLI logging i see that i am able to restrict the callerid when dialing out with asterisk. however, on the receiving phone, the callerid is still displayed. When i increment the logging of the pri with "pri set debug on span 1" on the CLI i also get the lower level debugging info from the pri.