similar to: ISDN Call Droping only for outgoing

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

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

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

Hi all, first off all - sorry for the cross posting - i did already posted this message to asterisk-dev - after that i realized that it isn't really a -dev related question - more a -users questions. So ignore it on -dev .... we have the following setup PSTN 3 PRI Lines <---> Asterisk (1.4.22) <---> Siemens HiCom <---> Bosch Integral The Asterisk Machine

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

Hi all. I am searching for a way to send SMS via our E1 PRI line. We are in Portugal and I have seen some Internet/TV/Phone providers (ZON for those who know it) who install normal phones with SMS support in landline. So I just found a page from PT (Portugal Telecom) stating that the SMC number is either 12999 or 129990 (

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

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

Hey, I wrote yesterday about PRI dropping, which turned out to just be a regular reset of unused B-channels. This time there's a real issue. As noted earlier I have an ISDN-30 connection, a Digium TE-121 with VPMADT032 echo cancellation. These are my configurations files: == /etc/zaptel.conf loadzone=dk defaultzone=dk span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 == ==

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

"RED: Loss of signal (LOS): The equipment shall assume "loss of signal" when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS." This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their

"RED: Loss of signal (LOS): The equipment shall assume "loss of signal" when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS." This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their

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

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

We are having an issue where DTMF is not being sent out right away and the tone duration is inconsistent. For a test we send a '5', then a second later we send a '9', and then five seconds later we send a '5'. If you look at the logs below you can see the first '5' is played right away, then the '9' comes in and gets queued, but it doesn't start

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

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|

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: