similar to: SMS in landline

hi all I have a PRI line which i have connected to my asterisk server. I am able to make calls to mobile no through my asterisk server, while i am not able to make calls to land line nos. This is strange. Where do u think the? problem is , is it from the service provider or? mis configuration of my asterisk. I am from India and using airtel pri lines. Below i am pasting you my configuration file

Hello, I have a problem, and I'm looking for you help. When I dial certain number my calls are getting dropped. I initiate the call, I hear IVR, then I am being transfered to operator, and then suddenly I get ISDN DISCONNECT message. I had this type of problem some time ago, and I thought it was a problem on the other end. But now this is a second time it occurred and I want an expert to

Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16.

Hi Guys, I am calling out 416-999-1111 on Channel 1 of PRI and then calling 416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/4169992222 -- Zap/2-1 is proceeding passing it

Hello List I'm trying to figure out what is wrong between my asterisk and my NEC 2400 pbx We have been trying to link them with a spare PA-24DTG from the NEC, I'm able to call an extension on the Asterisk, however the extension rings, and then immediatly hangs up I traced it back to the debug of the PRI on the Asterisk... I would appreciate if anyone could pin point what is wrong The

Told you it was too complex of an issue :-) I figured to do this in zapata.conf and all is fine now: transfer=no That was the magic two letter which was sending a request for RLT feature on the line. Set transfer to "no" and all worries gone. Thanks for the input everyone. -Bruce On Mon, Apr 12, 2010 at 10:10 PM, bruce bruce <bruceb444 at gmail.com> wrote: > Futher check

handled. So....what do I do? Thanks, MD =1=================================================== !! Invalid Protocol Profile field 0x11 -- Accepting call from '2004000' to '111' on channel 0/23, span 1 -- Executing NoOp("Zap/23-1", "Incoming call from Meridian1") in new stack -- Executing NoOp("Zap/23-1", " From number: 2004000|

I have a working system with asterisk 1.4.26.2 libpri 1.4.1 and zaptel 1.4.12.1 With a digium TE205p. I am trying to update to libpri 1.4.10.2. When I do, incoming calls work but outgoing does not. When I do this I "rm /usr/lib/libpri*" then just install libpri-1.4.10.2 as normal. I then do a make clean in asterisk and make distclean ,then configure, make and make install. I do

picture. I know the firmware on the Nortel is old, so I'm guessing that libpri is sending something that the Nortel does not know how to handle. Is there a way to dumb down what libpri sends? From everything I've read PRI is an evolving standard - and older devices may struggle with newer extensions/developments. (This might be very handy for users trying to talk to old pbx's.) Is

Hello, I have an Asterisk system in the UK using ISDN service from BT. My problem is that the called number is always passed to the provider with the Type Of Number declared as ?national? despite pridialplan (and prilocaldialplan) is set to ?unknown?. My questions are: - How can I find out what the effective value is for the above two variables without actually making a call? (I have very

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 all, I'm trying to insert a Rhino Ceros box equipped with a Rhino R2T1 dual-T1 card and running the latest version of Trixbox (2.4.2) between the central office and a Nortel Option 11. The switch at the CO is a DMS100. Basically, I'm taking the T1, connecting it to port 0 on the R2T1 card, and then connecting port 1 to the Nortel. (Actually a CSU and then the Nortel) We're running

I have been trying to sort this out for a while now but with no luck I have isdn <-> asterisk<-> pabx on a te205 incoming calls work fine outgoing calls seem to work fine but the call is dropped when answered I think it is to do with the line [May 8 17:51:55] WARNING[4762] channel.c: Unexpected control subclass '5' that is causing the problem but I don't know how to

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

I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on a CentOS 5 server. The server has a single TE110 card connected to a provider called Alestra in Monterrey, Mexico. Since we installed everything we have been having problems dialing certain numbers, those numbers always fail when dialed from Asterisk but if you dial from your cell phone they always go through. I once has a

I've a call center using IAX softphones provided by a third party. We've observed problems where the IAX phones seem unable to use our PRI trunks. A sample anonymized call is provided below with the PRI debug calls embedded. Any thoughts, comments or suggestions would be welcome. In anonymizing it, I preseved the format and number of digits sent. -- Accepting AUTHENTICATED

Hi everybody, I've connected Asterisk 1.2.5 (libpri 1.2.2, zaptel 1.2.4, Linux 2.6.13.2) to an Avaya-Tenovis PBX via a PRI/E1-line. Calls from SIP-phones via * to the PBX work fine. However, incoming calls to * only result in: -- XXX Missing handling for mandatory IE 24 (cs0, Channel Identification) XXX -- which seems to be an * problem, because a Windows-fax-machine works fine on a PRI

I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is

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

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