similar to: choppy music on hold - only on PRI PSTN

Hello, Anyone else experiencing problems with the latest spandsp (pre3) and last libtiff beta? I'm getting 8 bytes long file, with the TIFF header only during such connection: -- Accepting call from 'XXXXXXX' to 'YYYYYY' on channel 0/2, span 1 -- Executing SetVar("Zap/2-1", "FAXFILE=/tmp/foch.tif") in new stack -- Executing

I'm getting this intermittent problem, sometimes a zap channel gets stuck after a call. Below is a snapshot of the channel. Any ideas what can be happening? Name: Zap/1-1 Type: Zap UniqueID: 1056988772.10 Caller ID: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 68 WriteFormat: 4 ReadFormat: 4 1st File Descriptor:

I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the "Up" state, with asterisk consuming 100% of CPU: *CLI> show channels Channel (Context Extension Pri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None) (None) 1 active channel(s) *CLI>

Hi Group, actual i have a transcode problem and i have no idea to solve this. All my wav files are alaw encoded and i allow only alaw codec. But sometimes the WriteFormat is slin and if i recall the same number the WriteFormat is alaw for the Channel. Why the channel has sometimes slin and sometimes alaw? NativeFormats: 0x8 (alaw) WriteFormat: 0x40 (slin) ReadFormat: 0x8 (alaw) WriteTranscode:

Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P

Hi all, if i enter a "show codecs" at cli * response with: 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 << 4) MPEG-2 layer 3 32 (1 << 5) ADPCM 64 (1 << 6) 16 bit Signed Linear PCM 128 (1 << 7) LPC10

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

We have a variable set for each user/peer/whatnot that signals what the outbound caller-id should be sent as with our carrier. When someone dials a followme extension, this does not appear to be carried over for when the calls reach an outside caller, and we see the outbound caller-id being set as 'asterisk' vs the number desired. Has anyone else seen this, or found a way to

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

> Post your extensions.conf and what's on the CLI (asterisk -r) As requested: # cat /etc/asterisk/extensions.conf [incoming] exten => s,1,Answer() exten => s,n,NoOp(CallerID is ${CALLERID}) exten => s,n,NoOp(DID is ${DNID}) exten => s,n,Background(enter-ext-of-person) exten => 1625,1,Playback(digits/1) exten => 1625,n,Goto(digits/1) exten => i,1,NoOp(CallerID is

(this was also posted to the asterisk forum, but received no replies... Maybe someone here can help) I'm using the ices command to stream a conference to an icecast server. This is working nicely, for the most part, but the volume is very low. The streamed ogg vorbis audio is much quieter than what I hear in a SIP client, for example (on the same machine with the same audio hardware, of

hi list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name=

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

Hi, I'm very new to Asterisk and I have the following scenario. 1. Let's say I have a number of 1-222-222-2222 from my SIP service provider (VoicePulse). 2. I point my phone, Verizon wireless cellphone (1-111-111-1111), voicemail to the number provided by SIP service provider (1-222-222-2222). 3. I use another phone (1-333-333-333) to call 1-111-111-1111 and leave a voicemail message.

-- SIP/ vaso -e26c answered Zap/14-1 -- Executing DumpChan("SIP/ vaso -e26c", "") in new stack -- Executing DumpChan("SIP/vaso-e26c", "") in new stack Dumping Info For Channel: SIP/vaso-e26c: ============================================================================ ==== Info: Name= SIP/vaso-e26c Type=

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

Again - when Originate is run from dialplan, i get: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 192000) ReadTranscode: No When it's made with a call file (no matter how a call file is created), I see NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Please

Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted programmer_ted@hotmail.com P.S. Read to the very end. The original bugfix

Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1)