similar to: problem with dialing

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good day eveyone, I'm hoping that someone can help me. Perhapps i'm overlooking the obvious, in truth, I hope that I am. I've scoured the mailing list and google, and haven't come up with much. I have a Digium T400P thats been connected to a channel bank for testing for some time now. all has been well. I've now just had

Hi, After temporarily pulling the T1 cable out of our Asterisk box, we ended up getting a strange error messages even after the cable was plugged back in. [...] Dec 10 09:01:11 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:01:21 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown

Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs

Hi, I am trying to make a call from SIP to H.323 using chan_h323. Asterisk CVS-HEAD-03/10/05-10:08:22. As given in chan_h323 readme I compiled pwlib and open h323 versions 1.8.1 and 1.15.1. Call seems to be get connected but no audio path. I see following; -- AGI Script Executing Application: (DIAL) Options: (H323/YYYY#XX112422428@XX.103.19.91/XX112422428|60|HS(63840)) -- Setting call

I am using Asterisk CVS-HEAD-06/02/05-19:37:27. It seems that every call I made was duplicate. Jun 8 00:11:30 DEBUG[21733]: chan_h323.c:411 oh323_call: Placing outgoing call to 87874586, 101 -- Called 87874586 Jun 8 00:11:31 DEBUG[21733]: rtp.c:472 ast_rtp_read: RTP NAT: Using address 10.17.43.53:8000 Jun 8 00:11:32 DEBUG[21735]: chan_h323.c:1218 progress: Received ALERT/PROGRESS message

Hi, I have a queue that plays music when a call comes in. To be able to do that I need to Answer() the call first. After a timeout in this scenario the call should be transfered to an extension using a GoTo statement to the extensions context. The problem is that as soon as asterisk Answers the call it can not play a ringtone (or other tones) back to the original caller when executing a Dial

Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound & get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered

Hi Everybody, I am trying to use SIP (Sipura 2000) to connect to Asterisk which then dials out a local number using the Digium E100P. We have purchased the G729 codec licenses from Digium and loaded them into Asterisk successfully. However, the call drops immediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001

Hi to all, I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free ()

[I'm reposting this to the asterisk-users list, since it seems to be a bit more active.] Hello, I started messing with Asterisk few days ago, so my overall knoledge about it is still fairy superficial. I think I found an issue with MP3Player; it can be reproducted with this extension: exten => 6001,1,Answer exten => 6001,2,Background(blahblah) exten => 6001,3,Ringing exten =>

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

(I am reposting this question after a few months without a solution...) Hi all, I am trying to read a .txt file, with Hebrew column names, but without success. I uploaded an example file to: http://www.talgalili.com/files/aa.txt And tried the command: read.table("http://www.talgalili.com/files/aa.txt", header = T, sep = "\t") This returns me with: X.....ª X...ª......

On Fri, 15 Dec 2023 02:36:33 -0500 Mark Foley via samba <samba at lists.samba.org> wrote: > On Thu Dec 14 22:49:33 2023 Mark Foley via samba > <samba at lists.samba.org> wrote: > > > > On Thu Dec 14 19:27:29 2023 Matt Savin <matt at tegers.com> wrote: > > > I do not know who 'Matt Savin' is, but would he please reply to the list and not

Hi List, We have a problem in connecting ADS server from SAMBA. Whenever we try to contact the ADS server from the Samba its giving error as LDAP_NO_RESULTS_RETURNED. We have varified the code also, in the code also its giving three possibilities to raise that error message, I am pasting the portion of code here, File name: libads/ldap.c Function name: ads_server_info Code snippet

Hi, if i have data in the following time series format: time, amount, state 1 2222 A 1 333 B 2 45 A 2 77 B where states could be n and time periods t is there a package in R that would calculate the transition probabilities in a markov chain. for each t except t=0 to generate A B A B perhaps the best structure might

I am trying to connect asterisk to my PCI ISDN BRI card. I modify the modem.conf file and it runs just fine…giving the result [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem Driver) but then it gives the error saying… Sep 20 15:13:57