similar to: PRI cause code discrepancy

I am trying to retrieve the cause code of a outgoing call over a PRI where the number called is out of service. When an out service number is called I get a recording that the number dialed is not a working number. I see cause code 1 in in the CLI as soon as the call is dialed the Telco recording goes on for 30 sec. then hangs up. Any idea on how retrieve info that the called number is is

Well, it works for me .. YMMV. Yesterday I had a problem where I had a meridian talking to * via a PRI card, and from * to the pstn via an isdn30 link. The problem was that if the number was bad, or engaged then the meridian line simply dropped, not giving the operator any indication of what occurred. With much help from this list, I managed to construct a dialplan which solved our issues.

hello, i'm testing sending ISDN cause codes to customer pbx (test scenario for unallocated number) topology: PSTN-E1-AsteriskA-AsteriskB-SOMEPBX INVITE from SOMEPBX to PSTN AsteriskA sends to AsteriskB Status-Line: SIP/2.0 503 Service Unavailable X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 how can i resend HangupCauseCode from AsteriskB to

Hello, I have been running some tests to get a better understanding of PRIs and the HANGUPCAUSE variable and I'm not having any luck. I have tried calling disconnected numbers and the call is connected through to my extension and I hear the tri-tones. And it looks like HANGUPCAUSE is always 16 (AST_CAUSE_NORMAL_CLEARING). Am I doing something wrong, or am I just misunderstanding? Also,

> Hi Gareth Blades & Doug, > > Thanks so much for for the feedback. I have searched on lot of documents > but couldn't able to find clear answer regarding it. > > I hope you guys replies are very much help all in aterisk community. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Engineer, > > Debug Solutions > > Sri Lanka .

Hello all. Im try to make a SMTP Auth using Docecot SASL. Im use swaks for tests. Im store users in LDAP. As im understand for CRAM & DIGEST MD5 we need to store pass in a clear text?... Ok. mail: admin3 at domain.off userPassword: 123 <- Clear text What im do %swaks -a CRAM-MD5 -au admin3 at domain.off -ap 123 To: admin3 at domain.off === Trying mx.domain.off:25... === Connected to

Hi, I'm using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I've tested allow

Hi all. I have been reading the list archives and can't seem to find anything that relates to these errors. BackupPC_nightly fails to delete any files, and reports the pool and cpool at zero size. The nightly run as currently configured should be deleting a ton of files (only one hardlink) but it deletes nothing, and so the drive is now 100% and staying there. I have adjusted the conf

Hi, I was trying to use 2 of asterisk servers and interconnected, one of them as a peer to other sever (configured in sip.conf), so all the calls to server 1 will just be passed to server 2 (has PRI Card, TE 412P, only one PRI connected), i was sending calls to server 1 and that would send to server 2 and then dial out using Dahdi, but the problem that i got was the hangup cause codes, i was not

I want to say thanks to everyone for the help so far. I figured out a way to modify some AAH code that worked for me (well sort of). The line I modified is s,14 in macro-dialout-trunk. Then I just added a variable and passed it from 9_outside. I just have one last problem. This waits for an answer not ringing. So if the called party has a long ring to voice mail the call is dropped and goes

Hi all, on sourceforge.net I added the fixup release 0.5.4 of chan_capi-cm driver. The changes from 0.5.3 to 0.5.4 are: - fixed 'group' setting according to Asterisk defaults. - use SetCallerPres(prohib_not_screened) instead of CallingPres(32) for CLIR. - full CallingPres support added. - use mutex when debug/verbose messages are printed. - set dnid on incoming call. - catch errors in

Hi All, I have the following setup. Samba/LDAP PDC, Samba BDC, Samba member server, Win2K member server, 300 Win XP Client PCs. I can access the shares on the PDC from all Win XP clients. I can access the shares on the Win2K member server from all XP clients, I can't however access any of the shares on the Samba BDC or Samba member server from the XP clients. LDAP is working fine and

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

Hi All, the * is working rigth for incoming calls and internal calls, but when trying to call out we got hanged up. The hangup reason is AST_CAUSE_INVALID_IE_CONTENTS I've been searching in the mailing list archive as I thing that some thing similar happens to someone else but did not find. We are runnig asterisk 1.2.4 extensions.conf [default] exten =>

Asterisk 1.4.32 dahdi-2.3.0.1 Centos 5.5 Digium TE420 CAC channel bank (2) Cisco RVS4000 router Cox 50 Mbps/ 5 Mbps cable modem Flowroute provider codac G-711 90 % CPU idle callwaiting=no When there are 10-15 or more calls up the farend hears a callwaiting like beep every 3 to 6 sec. the duration of this "beep" is very short and would be no problem if it didn?t happen every few

Hey everyone, we are using several TE410 cards in a production environment that are connected to several operators PRI's and it works great.. For one of the operators we have seen some strange problems in cdr mismatches however. Our cdr's show phonecalls that are disconnected at a certain timelimit while the operators cdr's show the user has disconnected a lot earlier. I thought

I have two version 1.6 Asterisks running. One is a small hobbyist thing just at home, and the other is handling calls for several customers. On both, I have added the line exten => h,1,Set(CDR(hangupcause)=${HANGUPCAUSE}) to all relevant contexts. On my little hobbyist box this works perfectly; all calls have their hangupcauses recorded with cdr_adaptive_odbc and cdr_custom. On the

Thanks Dovid! Indeed looks a bug but regardless of this, this problem made me think that the HANGUPCAUSE could be used for this purpose with benefits. I couldn't find an explanation about when DIALSTATUS would actually be better. The HANGUPCAUSE was reworked in version 11 ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find someone actually stating it is a better

Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840 All works fine except for Asterisk->Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial("SIP/4053-0823dd48",

Looking for a router to connect to a 5/50 cable modem that works with Sip. A Crisco RVS4000 installed now has real problems with Sip, one-way audio and throughput not up to the WAN speed. No VPN needed, something affordable, $200-$350 US range. Every thing I looked at in that range had some reported problem except pfSense in a ATX box. Any recommendations or comments appreciated. thanks