[Asterisk-Users] X101P problems
All, I'm having some issues with an X101 clone. The machine is not plugged into the network right now, but I'll pull the configs and send them on shortly, but they are similar to the sample configs. Problem 1: ZAP cannot create channel For some reason, the cards hang after a call. It's pretty annoying since the only thing that will fix it is a restart of asterisk. There are two cards in the machine, both with their own, non-overlapping IRQ's. ztcfg and zap show channels show both cards properly configured and working. Inbound calls work properly, but outbound PSTN calls hang the interface. Here is the strange part, and the two problems may be related. The way outbound calls are originated is through a Sipura SIP adapter and an analog phone. During the call, as monitored at the command line, * shows the SIP call originating, the dial command for the ZAP interface being started and _the ZAP interface answering_, which seems to result in the originating station not getting a ring tone... After the call, it shows the SIP call terminated, but the ZAP channel remains unavailable until a * restart. Reloading the configs does nothing. Problem 2: No ring tone on SIP to PSTN calls When calling a PSTN phone from a SIP phone, there is no ring tone. Putting an 'r' at the end of the dial line results in a single ring only, under every circumstance. The * montering shows that the ZAP interface answers the call, which is wierd. Perhaps this has something to do with context? Should outbound calls be in a different context than incoming calls? Is it possible that * is answering itself? This would seem to be wrong as the remote number (my cell in this case) actually rings normally, there's just silence on the * side... Perhaps someone can shed light on my extensions.conf when I manage to get to it Problem 3: Random ringbacks I'm getting random ringbacks and just generally random rigs on the Sipura SIP adaptor. These seem to occur when the connection between * and the Sipura is disrupted, as well as at random times. Is there a way to fix or disable this behavior? It's very annoying. If anyone can shed any light on these problems, I'd be gratefull. I almost threw the box out the window today I was so frustrated... Thx. Chris. -- chris maresca senior partner - www.olliancegroup.com linux, up 27 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Cancellation Feature
The single most usefull tool that anyone outside telco consultants is likely to have is ztmonitor. If you do a ztmonitor [channel number] -v you will get a visual of the sound strengths and it's pretty easy to see when rx or tx are out of balance... Now, if only that would help fix the low-level static noise I have on the x100p, that would be great. Chris. On Thu, 22 Apr 2004, Rich Adamson wrote: I do feel the echo cancellation does need some work. Currently, other than listening to users, there is no way to benchmark or trouble shoot echo problems. Sure there are, it's just that 99% of the asterisk implementors don't have the test equipment to do it, and a good share probably wouldn't know how to do it if they had access to the equipment. We find that 2 to 3 out of every 20 calls will experience echo. While echo is a problem that naturally occurs from SIP - PSTN and vice versa, I am still baffled by the fact that the cancellation works randomly. When doing a zap show channel X, it will also report that the cancellation is still on. We experience the most echo with a T100P -- Adtran TA 750 FXO modules. While I understand these do not have impedence matching, I wonder to myself why echo cancellation works sometimes, and others not. Looking at Network util, processor util, and memory utilization, they do not provide any clear indication as to why /when it occurs. Not likely to have any impact whatsoever. Is there anything more that can be done to debug echo cancellation, and further are other users experiencing this random echo. I know it was discussed before, but the support folks at digium aren't able to offer anymore help. You've probably read enough from previous postings to know there are several different locations within an end-to-end voice call where echo can creap into a system. In very general terms, any place where an end-to-end channel incures a two-wire to four-wire conversion (whether done in hardware or software), echo can creap in due to lots of different reasons. Since asterisk provides us with lots of configuration choices, hardly any two systems are alike. Therefore, don't know that anyone is going to write * code anytime soon to correct something that can't be pointed to, etc. Someone mentioned they have echo on sip to sip calls (presumably the call was processed by a single * system). If they do, the problem is likely in the sip phone as there are no echo cancallation needs in that four-wire end-to-end call from an * perspective. I've got a fair amount of test equipment (and 20+ years telephony background), and am planning to assemble a document identifying some of the pstn issues, level settings, and other things impacting a reasonable system implementation. Unless someone wants to UPS a transmission test set to me quickly, the document won't be completed for several weeks. (The only test set I have access to will not be released for a couple of weeks due to classes, etc.) I'm also expecting these tests to point out a number of other transmission issues within asterisk that we'll get documented with real numbers, etc. Rich -- chris maresca senior partner - www.olliancegroup.com linux, up 17 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo Cancellation Feature
I've got a really cheap analog phone connected to a Sipura SIP adaptor, and have zero echo problems... Just static problems, but it may be related. Chris. On Thu, 22 Apr 2004, Brent Franks wrote: We have three Cisco 7940 SIP phones and 1 POTS phone connected to our * server with TDM10B fxs card. Our * server is connected to the pstn with 3 X100P cards. We have similar echo problems but only on our SIP phones. We do not have any echo problems with the POTS phone. We just purchased a Polycom IP 500 SIP phone to test but I expect similar echo problems. Don't expect the IP 500's to do anything as you stated, we are using these now and are having the problem I described earlier with these phones. - B -- chris maresca senior partner - www.olliancegroup.com linux, up 17 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)
I'm not surprised. My wife is an editor at on of CMP's more cluefull publications, and, even there, she's fighting against general stupidity evey day. Most journalists, esp. tech journalists, are pretty clueless. Chris. On Sat, 17 Apr 2004, JR Richardson wrote: * Brethren, It's a sad day in our community. Please join me in a moment of silence for the death of responsible journalism. Silence.good enough. This article goes on to tell about Pingtel's announcement of forming the first open source community aimed at creating SIP based servers. http://www.networkmagazine.com/shared/article/showArticle.jhtml?articleId=18 900066classroom= Yipeee! I am embarrassed and appalled with the lack of recognition or mention of all the tremendous work in this already existing community. I'm writing a letter to the editor and encourage all of you to do the same. Send your comments to [EMAIL PROTECTED] Mark Spencer, I caught your presentation at the Linux-Kongress http://graphics.cs.uni-sb.de/VCORE/recordings.html . I want to personally thank you for validating all that I have been doing to promote Asterisk to anyone and everyone who will take the time to listen. When I talk about it and show off working systems in action, people get excited and are quite impressed. Having your 30 min presentation to go along with my demo increases my credibility 10 fold. Thank you all for contributing to this great community. JR ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect Problems
They just updated their software and that seems to have resolved the DTMF issues, at least for me. Chris. On Tue, 13 Apr 2004, Andrew Kohlsmith wrote: 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. Works just fine for me. Don't send in-band DTMF if you're not using the alaw/ulaw/slinear codecs. It won't work. Regards, Andrew -- chris maresca senior partner - www.olliancegroup.com linux, up 8 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Trunk to PSTN (voicepulse) questions...
All,
I've almost got my Asterisk PBX setup, but I've having some problems with
the VoicePulse IAX trunk.
On outbound calls, when dialing a PSTN number through the IAX2 trunk,
music on hold (moh, using the m option in the dial command) does not work.
The console states that stop sound on IAX2 channel. Ring works, but
only without the r option. MOH works when trying to dial a non-PSTN
terminated IAX2 calls (e.g. a softphone). I've read that with SIP
connetions, the originating line is not held open by the PBX, so the can
be no timing sync with the client, but I don't know if that's also the
case here.
The setup I have is:
[sip softphone Xten] == [ * ] == [IAX2 VoicePulse Trunk] = [PSTN Number
(SprintPCS Cell)]
The relevant iax.conf sections are:
[voicepulse]
context=voicepulse-incoming
dtmfmode=rfc2833
secret=mysecret
auth=md5
type=user
host=gw5.voicepulse.com
[voicepulse-peer]
qualify=yes
trunk=yes
dtmfmode=rfc2833
secret=mysecret
auth=md5
type=peer
host=gw5.voicepulse.com
My extensions.conf has:
TRUNK=IAX2/[EMAIL PROTECTED]
exten = 15,1,Playback(transfer)
exten = 15,2,Dial(IAX2/ckm,20,rt)
exten = 15,3,VoiceMail(u${EXTEN})
exten = 15,4,Hangup
exten = 15,103,Dial(${TRUNK}/1411212,30,t)
exten = 15,104,VoiceMail(u${EXTEN})
exten = 15,105,Hangup
Any ideas, bug?
Thx.
Chris.
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Re: [Asterisk-Users] IAX2 DTMF Problem
I have the same problem, got this from VoicePulse today: Chris, Thank you for contacting VoicePulse. Our engineers are aware of the DTMF problem and are working to have it resolved as quickly as possible. Please reply directly to this email if we can provide any additional assistance. Regards, VoicePulse Customer Support On Fri, 9 Apr 2004, Steven Critchfield wrote: On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and everything works great, but I am not sure how to attack this. I found a few items referring to the same issue in the list, but I didn't find any answers. If this is a bug I will create a report on the bugtracker, but I would rather make sure that I am not just completely dense and not seeing the easy answer. I'm trying to replicate the issue with NuFone. CVS from 2004-04-04 stable branch. Is this in the extensions.conf file or a agi? either way, maybe you should make sure you Answer() the call before anything else. After that and a clarification of where youa re looking for the DTMF it may be easier to answer your question. -- Steven Critchfield [EMAIL PROTECTED] -- chris maresca senior partner - www.olliancegroup.com linux, up 3 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mpg123 issue and solution
All, I've been setting up an Asterisk PBX over the past few days and I stumbled across a problem with mpg123 that I have not seen documented anywhere. If you put mp3 files into your mohmp3 directory and these files have ID3v2 tags, mpg123 will throw an error message Found new ID3 Header, regardless of the -q flag. This, in turn, will cause Asterisk to crash (yes), although it's a soft crash (exits cleanly). It took me forever to figure this out, since the default mp3 and everything else was working fine. And the lack of any meaningfull error messages made diagnosis even more difficult My work around was to open the file in WinAmp and remove the ID3 tags entirely. mpg123 and Asterisk were both happy and there was much rejoycing. It might be a good idea to move away from mpg123 as it is no longer supported and there are bound to be more problems like this. MAD seems to be what everyone is migrating to... At the very least, not hardcoding a player into the codebase would probably be a good idea (if it is hardcoded, I couldn't find a config file for it...). On a seperate note, if anyone is have problems registering SIP softphones, like the most excellent xTen X-Lite, try commenting out 'bindaddr' in sip.conf. That allowed me to register and get everything working. Finally, if anyone has any ideas about how to improve IAX voice quality, I'd be happy to hear them. Everything is hearable, but there are an awfull lot of clicks and pops in the background. The machine is a Celeron 500 with 128mb of RAM and Gentoo 1.4 (w/latest gentoo updates) and Asterisk 0.7.2. I'm the only one using the machine ATM and it's about 2 ft from my desk with a dedicated hub... There is no telephony hardware install currently. SIP quality seems slightly better, and I've tried almost every WinIAX client out there, they're all pretty much the same. I'm using the soundcard in my laptop and a Plantronics headset. Playback of the IVR prompts generally sounds good, but inter-client communication is noisy. Thx. Chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 issue and solution
Apperently this is only a problem with v59r of mpg123. Older version don't have a problem. Chris. On Tue, 6 Apr 2004, Justin Carlson wrote: I have a suse 8.2 installation of mpg123 and I have no problems with the id3 tags -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Tuesday, April 06, 2004 11:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] mpg123 issue and solution Hi! If you put mp3 files into your mohmp3 directory and these files have ID3v2 tags, mpg123 will throw an error message Found new ID3 Header, regardless of the -q flag. This, in turn, will cause Asterisk to crash (yes), although it's a soft crash (exits cleanly). It took me forever to figure this out, since the default mp3 and everything else was working fine. And the lack of any meaningfull error messages made diagnosis even more difficult My work around was to open the file in WinAmp and remove the ID3 tags entirely. mpg123 and Asterisk were both happy and there was much rejoycing. See also: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf (as well as the Wiki Asterisk FAQ section concerning variable bit rate) It might be a good idea to move away from mpg123 as it is no longer supported and there are bound to be more problems like this. MAD seems to be what everyone is migrating to... Indeed mpg123 is known to be the cause for many problems. Cheers, Philipp Finally, if anyone has any ideas about how to improve IAX voice quality, I'd be happy to hear them. Everything is hearable, but there are an awfull lot of clicks and pops in the background. This is probably due to the IAX software phone that you are using (and its underlying library). On * server to * server connections this shouldn't be the case (if yes: try to enable/disable the jitter buffer, see other mails here). -Philipp- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- chris maresca senior partner - www.olliancegroup.com linux, up 1 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] registration failure
Comment out bindaddr. That did it for me. Something to do with SIP and DNS resolution, which is a problem if your PBX is on a private network and has no DNS entry HTH, Chris. On Tue, 6 Apr 2004, Roger wrote: I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.22.1' Apr 6 14:46:34 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.22.2' Apr 6 14:46:50 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.22.1' The ips in the 22.1 and 22.2 are 2 Cisco 7940 phones running SIP image 6.3. The host 22.254 is the dhcp/tftp server/asterisk box When I attempt to dial extension 2000 from ext 2001 I get the following... Apr 6 14:51:57 NOTICE[1116957488]: chan_sip.c:5337 handle_request: Failed to authenticate user 2001 sip:[EMAIL PROTECTED];tag=000a8a490a3d00056f69e7bc-18df5ff6 Apr 6 14:52:04 WARNING[1116957488]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Response) sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) ;bindaddr = 192.168.22.254; Address to bind to (all addresses on machine) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=cisco; Password for device ;host=192.168.22.1 ; This host is not on the same IP addr every time host=dynamic context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001]; Duplicate of 2000, except with different auth data type=friend username=2001 secret=cisco host=dynamic ;host=192.168.22.2 context=from-sip mailbox=101 [2002]; Duplicate of 2000, except with different auth data type=friend username=2002 secret=cisco ;host=192.168.22.3 host=dynamic context=from-sip mailbox=102 extensions.conf [2000] exten = 2000,1,Dial(SIP/2000,20) [2001] exten = 2001,1,Dial(SIP/2001,20) [2002] exten = 2002,1,Dial(SIP/2002,20) *CLI sip show users Username Secret Authen Def.Context A/C 2002 ciscomd5,plaintextfrom-sip No 2001 ciscomd5,plaintextfrom-sip No 2000 ciscomd5,plaintextfrom-sip No *CLI sip show peers Name/usernameHost Mask Port Status 2002/2002(Unspecified) (D) 255.255.255.255 0Unmonitored 2001/2001(Unspecified) (D) 255.255.255.255 0Unmonitored 2000/2000(Unspecified) (D) 255.255.255.255 0Unmonitored And from the SIPmac.cnf I have ### New Parameters added in Release 3.0 ## # Phone Prompt (The prompt that will be displayed on console and telnet) phone_prompt: SIP Phone ; Limited to 15 characters (Default - SIP Phone) # Phone Password (Password to be used for console or telnet login) phone_password: cisco ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: 2001 I really really I'm close - just stuck on this last problem. Any suggestions would be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- chris maresca senior partner - www.olliancegroup.com linux, up 1 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
