similar to: Delays while playing a message

Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%

I have some DID numbers that come into Asterisk via PRI, then connect to a Panasonic DBS PBX via PRI. Outbound calls from the PBX go out via PRI to Asterisk, then out to the Telco's PRI. The sync LEDs on my PBX show that it is synced to Asterisk via the PRI. I have users complaining about random hangups. What is the best way toi approach finding the issue? ref: zttest looks good. ./zttest

Guys. I have an issue with a te110p card and also some tdm04b cards on the same system: Zttest returns this for the tdm04b cards: [root@mollendo ~]# /usr/src/zaptel-1.2.4/zttest 38 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.000000% 8192 samples in 8192 sample intervals 100.000000% 8192 samples

I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

Another request make me test my t1 card, which has no quality problems, but all that I get is: [root@SIP2-MI zaptel-1.2.1]# ./zttest Opened pseudo zap interface, measuring accuracy... 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

I just needed to test a dialup modem connection (don't ask) and I had a modem connected to a TDM card (FXS port) which then dialled out via a E1 PRI on a TE4xxp card. See my log below: atdt0198xxxxxx CONNECT 36000 V42bis ** Dial IP ** Username: xxxx Password: Entering PPP Session. IP address is xxx.xxx.xxx.xxx MTU is 1500. OK ATH0 OK ATDT0198xxxxxx CONNECT 31200 V42bis ** Dial

Hi Steve and all, 1. Faxing from asterisk back to the same asterisk (from one Zap channel to another) doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the receiving side needs CNG in order to switch to fax extension with rxfax. 2. This is probably the reason why J2 and our UC don't recognize incoming fax. Thank you. Alex Zarubin Webley Systems

Hello, The v1-0_stable cvs release doesn't include the recent change ('poll' instead of 'select') in channel.c. Will it end up there any time soon, or we need to use cvs head to pick up this change? Thank you. Alex Zarubin Webley Systems -------------- next part -------------- An HTML attachment was scrubbed... URL:

With multiple inbound PRIs (and hunting across them) coming to multiple [asterisk] servers it is important to be able to do administration, i.e. control which PRIs in the same hunt group take (and which don't take) calls from telco at any given period of time. Our pre-asterisk platform uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance'

Please, try exten => 99,1,Wait,1 exten => 99,2,Record,/tmp/pcmfile:pcm exten => 99,3,Wait,1 exten => 99,4,Playback,/tmp/pcmfile exten => 99,5,Wait,1 exten => 99,6,Record,/tmp/voxfile:vox exten => 99,7,Wait,1 exten => 99,8,Playback,/tmp/voxfile (put your own extension). Pcm recording is OK, playback is OK. Adpcm recording is noticeably worse. Adpcm playback is very

Hi, It is important for some applications (voicemail and the like) to control play volume and speed if instructed by the caller. The question is: how to do it the right way with asterisk. As far as volume - adding txgain and rxgain to the call structure[s] and setting gains from there (instead of taking global values from config files) looks like an OK solution. As far as speed - the easy way

Hi, We have no problems sending to HP and Panasonic fax machines in the office. We do have problems when we try to send faxes to services supporting fax, i.e. J2 or our UC platform. The receiving side doesn't recognize fax. To send a fax we drop into /var/spool/asterisk/outgoing: Channel: Zap/g1/<fax number> MaxRetries: 0 WaitTime: 20 Context: webley_txfax Extension: txfax_ext

I am having trouble with zttest on a Tyan board, dual AMD Opteron's on FC3. Tried different kernels, no IRQ sharing, everything looks in order. My zaptel modules load fine, but if I run zttest, it just hangs. Below is the strace output and you can see where it stops. Anyone have any ideas? [root@asterisktest zaptel]# strace ./zttest execve("./zttest", ["./zttest"],

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

Hi, I have a TDM400P running with only one FXO port running on a VIA EPIA MS10000 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it hang and when I interrupt it with Ctrl-C that is the result: ?anyone have some idea about why isn't working? Some additional info: # /usr/src/zaptel/zttest -v Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best:

> They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and > 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff

Is there a way to play adpcm, 6KHz in asterisk? If yes, where can we get this codec? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/b8be2453/attachment.htm

Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with

Our asterisk system gains access to the PSTN through a voip provider. We have no digium or other telephony hardware in our system. Do the zttest results still matter to us? Our results were as follows: --- Results after 1007 passes --- Best: 100.000000 -- Worst: 99.780273 -- Average: 99.975763