similar to: T1 PRI

>Freddi Hansen wrote: >> with boards from Aculab, we are replacing Aculab boards with Digium >> boards BUT we would need more >> Digium boards IF we could use both Digium and Aculab cards in the same >> server. The reason being that >> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards >> in the servers that must support >>

Hi all, I'm looking for a way with any asterisk-version with TE410P (cpe EuroISDN, Q931) for sending an INFORMATION ELEMENT KeypadFacility, eg. *87, during a connected call to the PSTN switch. Are there existing functions in asterisk to generate & send such IE ? If not what existing modules would be best to derive from? TIA, Bruno -------------- next part -------------- A non-text

Does anyone know how to monitor * to see if they are receiving timing slips on a span connected to a T100P card? I am seeing b-channel restarts quite often and also getting "No D-channels available" warnings from time to time. Yesterday I had all the b-channels crash during a MeetMe Conference. Not good! This PRI is connected to an Avaya Definity PBX that is onsite and located in the

Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

While testing a problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon this problem: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1210770512 (LWP 11346)] 0x002e3fe1 in pri_release_timeout (data=0x93a0478) at q931.c:2589 2589 q931.c: No such file or directory. in q931.c q931.c is in libpri, function

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

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

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, My configuration is pri----*(te405p)---iaxclient. My * version is 1.0.7 running on tyan dual opteron board. I have several problems. 1) inbound echo For outbound call(iaxclient-->pri), there is almost no echo. But for inbound(pri-->iaxclient), I can hear distinct echo. Can Sangoma a104 or digium te406p help this problem? 2)Today i received te406p. I know T1/E1 jumper. But how can i

Hi all, I've been trying to add a custom mysql field to my CDR's, but I must be doing something wrong. I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I add: exten => h,1,Set(CDR(q931)=${HANGUPCAUSE}) This extension is executed, I can see it in the asterisk console. I have added a new column in my MySQL database called q931. However, the new field does not show up in

Hello, I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo cancellation. Every 30-60 minutes I experience PRI dropping. @@@ /etc/zaptel.conf: loadzone=dk defaultzone=dk span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 @@@ @@@ /etc/asterisk/zapata.conf [channels] switchtype=euroisdn usecallerid=yes group=2 signalling=pri_cpe context=incoming channel => 32-46

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

Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-00000002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-00000002 -- DAHDI/i1/6463279153-2 is busy -- Hungup

Hi Guys, We have a problem with Caller ID not being displayed. I want to test everything to see where the problem is with the incoming Caller ID and why it's not displaying. I am tracking this down to "Presentation prohibited of network provided number" even though the Caller doesn't use *67 and even though they haven't asked their provider to block their CLID for outbound.

Hi Guys, Can q931.c be re-compiled using gcc or something else without the need to re-do the whole libpri? Some changes were made to q931.c and I want those to be reflected in .a .o .so .lo files as I think those are the files read by Asterisk vs the .c file. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk setup with outgoing calls not completing and requiring an Asterisk reset to 'unstick' span 1. Sorry this is a bit long but I'm completely out of my depth :-( This system has been in use for some while and I recently upgraded it to asterisk 1.4.24, zaptel 1.4.11 and libpri 1.4.9. I didn't change

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

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 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