similar to: 2BCT last mile... Hopefully

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

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.

I'm trying to get "blind transfer" from an incoming DAHDI line to an external number to work on an * 1.6 install using a T1 from XO. The documentation is very "distributed" and incomplete, so while it's not working, it's definitely more likely my error somehow. Couple questions if anybody is out there who even knows what TBCT is. 1) Is this even

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. 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 (

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 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

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

I've freshly installed CentOS 5 32-bit on a pc, performed a yum update and yum upgrade. I then did a yum install of mysql. I finally picked up fresh copies of phpmysql and bugzilla and put both in my /var/www/html directory. mysql is running. I am able to use mysql -u root to get into it, and phpmysql is able to get it, too. After runs of checksetup.pl to ensure all required

I have a new Dell PowerEdge 2950 running CentOS 5.0 out-of-box and a Dell MD3000i. I am new to iscsi and, with google and included documentation, am having a heck of a time trying to get the RAID volumes I have created on the 3000i to be seen by the OS as usuable drives. I have printed out SMcli and iscsiadm documentation. I have asked on the linux-poweredge at dell.com site, too. Many

I have not been able to get two B-channel transfer to work on DMS100 PRI. I consistently get the following errors: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN ERROR: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: OPERATION: RLT_OPERATION_IND [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ERROR: RLT Not Allowed I have tried on two

Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation.

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 I have done everything richard told to do ECT . below is my trace, anyone can help ? -- DAHDI/i1/09123278669-4 answered DAHDI/i1/88050048-3 -- Native bridging DAHDI/i1/88050048-3 and DAHDI/i1/09123278669-4 PRI Span: 1 Adding facility ie contents to send in FACILITY message: PRI Span: 1 ASN.1 dump PRI Span: 1 Context Specific/C [1 0x01] <A1> Len:11 <0B> PRI Span: 1

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

Hi, I'm trying to figure out the cause of a soft lockup I experienced: Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s! [asterisk:32029] Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[<c046e7fe>] CPU: 0 Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c Mar 29 09:38:24 pstn1 kernel: EFLAGS:

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

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

I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian Option 61C. Calls either way drop with error "Channel 0/23, span 1 got hangup, cause 100". Can anyone offer insight into the cause and solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading matching zaptel & libpri, put the problem is identical). For testing, I tried a call from the

CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? It appears that RealTime for the extensions.conf file is on a context by context basis, but you have to create each new context in the extensions.conf file then add a "switch => Realtime" line (then reload). I want to be able to add phones without having to edit any files.