Re: [Asterisk-Users] Re: IAX2 trunking on one side only.
I sent this yesterday, but for some reawson it did not go through. Yes, ASTERISK1 = 2x TDM400P ASTERISK2 = 3x X100P I still cannot get it working past that. Is there something screwey with the wcfxs drivers and Linux? - Original Message - From: Louis-David Mitterrand [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 07, 2003 1:59 AM Subject: [Asterisk-Users] Re: IAX2 trunking on one side only. On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote: Hello, I have searched google, read everything on the mailing list, read /usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on the IRC channel and I cannot figure out what is wrong with my IAX2 trunk. Only asterisk2 of an ASTERISK1--LAN--ASTERISK2--PSTN will use IAX2 trunking. If I do an iax2 show trunk on asterisk1 it says 0 calls on trunk Do you have a zaptel device on each side? AFAIR zaptel timing is required for trunking to work. -- If Galileo is the spark that lights up the gas giant Jupiter, turning it into a second sun, that will be the last straw. We will then have no choice but to make safety the number one priority at NASA. -- falsification on /. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200
Yep, we have 8 SNOM200's in an installation and none of them have a usable speakerphone. There is something frelled up with the voice detection on those phones when using speakerphone. We talked to kevin in technical support for the American Distributor and he told us to try the latest beta firmware. Upgrading had not helped at all, the other thing he hinted to was a new bootloader that overclocks the board in the phone and that might sometimes help. I have not called him back yet to tell him that upgrading the bootloader and firmware has not helped. I will update you if any changes, or you can be really cool and call him and let me know if he can fix your problem? - Original Message - From: Mark Evans [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 07, 2003 8:22 AM Subject: [Asterisk-Users] Snom 200 Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IAX2 trunking on one side only.
Yes, ASTERISK1 has 2x TDM400P and ASTERISK2 has 3 x100P cards in it. I'll try to dork with the timer, but as long as wcfxo or wcfxs is loaded shouldn't that take care of these issues? - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 07, 2003 4:02 PM Subject: Re: [Asterisk-Users] Re: IAX2 trunking on one side only. Louis-David Mitterrand wrote: On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote: Hello, I have searched google, read everything on the mailing list, read /usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on the IRC channel and I cannot figure out what is wrong with my IAX2 trunk. Only asterisk2 of an ASTERISK1--LAN--ASTERISK2--PSTN will use IAX2 trunking. If I do an iax2 show trunk on asterisk1 it says 0 calls on trunk Do you have a zaptel device on each side? AFAIR zaptel timing is required for trunking to work. We have to rename Zaptel timing to Asterisk timer, which is more correct since there are several ways of getting a timer to work, only one of them is by using Zaptel cards. http://www.voip-info.org/tiki-index.php?page=Asterisk+timer /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No communication channel
Whenever I have had this problem it was certainly a SIP/Firewall issue. Calls will go through but audio (RTP) will not go through. Second, using iconnect I have to add an option r (I think) to my dial command in extensions.conf to make asterisk send ring back to the originating phone, because I think something gets messed up between asterisk and iconnect's sip messages. I realise none of this directly answers your question, but I hope I have given you enough source material that you can start looking in the right direction. - Original Message - From: Lal, Deepak (Contractor) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 07, 2003 3:25 PM Subject: [Asterisk-Users] No communication channel I have following setup: AnalogPhone_1--TDM400P--Asterisk---SIP---[Softswitch]POTS-AnalogPhone_2 I can call from AnalogPhone_1 to AnalogPhone_2 and all is fine. When I call to AnalogPhone_1 from AnalogPhone_2, AnalogPhone_1 rings BUT I hear no ringing tone AND when someone picks up AnalogPhone_1, there is no sound and parties on both end cannot hear each other. Seems that no communication channel is open! Any suggestions? Thanks - DL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk + dual phone lines + cisco + backup
I haven't done this, but Use the Linux high availability project's heartbeat software Configure both servers to watch the same IP. T each of the incoming PSTN lines to dual x100p cards (one on each server) Configure one asterisk server to pick up the line on incoming calls a few seconds later than the other. Configure the dialplan to do something like this Check and see if the sip phone is available, if it is not we can presume the other asterisk server has taken over the ip and we send the call over to it through another dedicated network interface. Otherwise just send the call through to the sip phone. On the backup asterisk server we configure it in a similar manner. If it does receive the call (because the other asterisk server never answered the line) then it will check and see if the sip phone is available if so it send it off to it. If not it will presume the ip has been taken over by the other box and send the call over to it. Going the other way it will be similar... Configure both sip.conf to latch onto the shared IP between the two asterisk servers. When the sip phone generates a call it will automatically go to the current active asterisk switch. The asterisk switch will check if the Zap/? is available and send the call out if it is. If it is in alarm it will send the call to the other asterisk switch for routing out to the PSTN. Configure both servers identically as far as asterisk goes and write some script that will scp or rsync the /etc/asterisk directories and restart/reload asterisk after every time you make changes to the configs on both servers. Use NFS or SMB/CIFS to have only one /var/spool/asterisk/voicemail directory. And if I am not totally on crack, I think that will work. But gotta ask is it really worth doing all that for two PSTN lines? If it were a really important site like a hospital or 911 call center I could understand, but this is just your home! If it works let me know, need any help I would be happy to work on this with you (I think I could learn alot). - Original Message - From: Ling C. Ho [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 07, 2003 4:41 PM Subject: Re: [Asterisk-Users] asterisk + dual phone lines + cisco + backup Dragan Mickovic wrote: I have couple of questions about the following. Currently I have 2 phone lines going into my house, and I would like to have both of those coming into asterisk. I also want to have a backup asterisk, so here are the main questions (I am knew to this so I apologize if I ask something stupid): - Is there a dual FXO card available from digium or do I need 2 x single FXO (if this is the case then I'll need 4, 2 more for the backup asterisk). - Since I want both asterisk boxes to have the same extensions, is there any internal procedures that do this, or maybe procedure for getting extensions from SQL? .. as a worst case I can always rsync the data :) - I have a cisco 1750 with 2FXO, is it possible to use it, has anybody done it, and if it can share some sample configs? I have setup asterisk to work with a Csico 2600 with FXO using SIP, and Xlite. My extension.conf is like this: exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED]) ; Please hold while... exten = _XXX,2,Playback(transfer,skip) ; Please hold while... exten = _XXX,3,Macro(stdexten,43878,SIP/[EMAIL PROTECTED]) ; (but skip if channel is not up) * 10.0.0.150 is the cisco box ip * 10.0.0.249 is the ip for my asterisk box * 43878 is my internal centrax extenstion * we need to dial *9 to get outside line, this is being done at the Cisco * 3878 is my softphone extension On my cisco: ! dial-peer voice 924 voip incoming called-number 3878 destination-pattern 3878 session protocol sipv2 session target sip-server codec g711ulaw ! dial-peer voice 927 pots destination-pattern ... port 1/1/1 forward-digits all prefix *9 ! sip-ua sip-server ipv4:10.0.0.249 I have a POTS dial peer setup on the cisco box to dial through the FXO if I make a 7 digit calls from my softphone. Also, there is a VOIP dial peer to send incoming calls to my Asterisk box. Basically, I can call 7 digit numbers from my softphone (should be easy to expand that to 10 digits), and it will call an outside number directly for me. If someone need to reach me, they call the number of one of the voice line connected to the FXO card on the Cisco, get a dial-tone, then dial and extension to ring my soft phone. I think you can configure the cisco to directly route an incoming call to one of the voip destination, but I don't have a set up ready to show. This is just a simple test I did a few weeks ago, doesn't cover a whole lot, but it's possible to make use of the FXO card on the cisco if you already have one. ... ling - Having both asterisk boxes using the same lines at the same time, is there anything that
[Asterisk-Users] IAX2 trunking on one side only.
Hello, I have searched google, read everything on the mailing list, read /usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on the IRC channel and I cannot figure out what is wrong with my IAX2 trunk. Only asterisk2 of an ASTERISK1--LAN--ASTERISK2--PSTN will use IAX2 trunking. If I do an iax2 show trunk on asterisk1 it says 0 calls on trunk to asterisk 2 (show channels does show the calls). If I do iax2 show trunk on asterisk2 it says 7 calls on trunk to asterisk1. I am using GSM and when I look at the traffic using iptraf with 7 calls active from asterisk1 (analog phones TDM400P) to ASTERISK2 Milliwatt() I see asterisk1 transmiting at a little more than 30k above what asterisk2 is transmitting. I have tried peer/friend, notransfer(?),registration/no registration and nothing about the trunking issue changes. Here is my config, some please tell me what I am doing wrong. ASTERISK1 iax.conf [anistone] type=peer (friend/peer) host=172.16.1.5 (with and without this statement) secret=test2 context=local2 (with and without this statement) trunk=yes extensions.conf exten = 61,1,Dial(IAX2/gateway:[EMAIL PROTECTED]/[EMAIL PROTECTED]) ASTERISK2 iax.conf [gateway] (I have tried it with this also named anistone) type=peer (friend/peer) host=172.16.1.232 (I have tried it with and without this statement) secret=test context=anistone (with and without this statement) trunk=yes extensions.conf exten = 60,1,Milliwatt() Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH 43017 Phone: 614-798-9106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk system lock
- Original Message - From: mattf [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 04, 2003 4:51 PM Subject: [Asterisk-Users] Asterisk system lock Hello, In the last week I've been getting a lockup about every 2 days. during the lockup the people that are on the phone can keep talking, but noone can initiate any kind of call internal or external. I went into the manager interface and tried a Action: Hangup and Manager gave me a Success message back only to see that the Zap channel was still active in the show channels screen. When I eventually do a stop now it takes about 20 seconds before it finally stops and it says segmentation fault. then I start it back up and everything works fine. Does Asterisk need to be stopped and restarted regularly to not crash and freeze? Any ideas? I am using a CVS from October 1st and have the following config: Digium quad t1 card (400) P4 2.8GHZ, 2GB RAM, SCSI RAID RedHat 9 2 T1's hooked up 50 Grandstream phones set up. Any help would be greatly appreciated. Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I have noticed a similar problem, sometimes it was a configuration issue (nothing major, syntactical) other times it was the version of cvs. I do not have hard facts to give here, but I have seen things similar, but not as easily reproducable as yours. Just today I did a restart gracefully no one was on the phone and it did not restart. The only command I could execute was exit, though I could always get back in doing asterisk -vr. I finally had to run around and make sure everyone was off the phone then kill the asterisk process and restart it and it worked fine. During the kinda fake hang no new calls could be established, after I restarted asterisk it was fine. As far as do you have to reboot asterisk every so often, my answer is no. If you are experiencing (sp?) problems like that you have something screwey. While using the prototype TDM400P's I did have to reboot regularily, and if I have a TDM400P with bad modules in the asterisk server it causes considerable instability. Other than that I have never had to regularily reboot an asterisk server. The only reason they get restarted now is for hardware change reasons. I have one server up for about 2 months now without a single problem. Hope some of this info is helpful. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Software FAX
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy over a bunch of include files to
/usr/local/include
2) build/install spandsp
3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source tree.
4) move Makefile.patch from oncall to apps/ dir in asterisk
5) patch the makefile
6) edit the makefile and remove all references to steve's home dir to make
it point to my spandsp source directory.
7) rebuild/install asterisk
8) Create a dir incoming/ in /var/spool/asterisk
9) edit extensions.conf and add the following line to the incoming call
contexts I have set up.
exten = fax,1,RxFAX(/var/spool/asterisk/incoming/${CALLERIDNUM}.tif)
10) create a script that emails me the tif files every time they are
received in incoming/ and delete them.
#/bin/sh
cd /var/spool/asterisk/incoming
for X in *.tif
do
if [ -f $X ] ; then
mutt -s FAX from $X -a $X [EMAIL PROTECTED]
/dev/null
rm $X
fi
done
11) Add a cronjob to run my script every 5 minutes.
*/5 * * * * /usr/sbin/mailfax
12) Test and enjoy.
To send a fax all I have to do is
1) Get the .tif file on the server somewhere
2) Put a file sample.call in the /var/spool/asterisk/outgoing/ directory and
it looks like this...
Channel: Zap/3/7989106
Application: txfax
Data: /root/fax.tiff
3) Asterisk will send it or keep trying until it send it as soon as I :wq
the file in vi.
Pretty simple, I hope this helps someone else.
Brian J. Schrock
Anistone Technologies, LLC
6926 Avery Rd.
Dublin, OH 43017
Phone: 614-798-9106
FAX: 614-798-9106
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[Asterisk-Users] Artificially Limiting IAX Calls
Everyone, Can I artifically limit the amout of IAX calls going out of an asterisk server... I am worried that if every call requires x amount of bandwidth and their internet link is only as big as 2x what happens when the third call comes up. So far I have seen it just starts up and will kill the other two calls. How have other folks on here dealt with this issue? Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH 43017 Phone: 614-798-9106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
I'll give it a shot, but that stuff is a little out of the realm of my knowledge. I remember a BIOS black magic web page from this past year on slashdot, but I don't remember anything about it. Do you have any advice on which options I should specifically look at? To add to my previous e-mail, I have disabled all on board hardware that was unneccessary on the kt4vl, audio, usb, serial ports, parrallel ports etc. I would be interested in tracking down exactly what caused the problem, I still have the board lying here, with the exact same memory and cpu that was in it. I look forward to some options to try. - Original Message - From: Dana Dominiak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 17, 2003 6:10 PM Subject: [Asterisk-Users] (no subject) Brian Hello, I resolved my echo issue using grandstream/estara etc etc Brian sip phones and wcfxo interfaces from digium. I swapped out my Brian via kt400 based msi kt4vl motherboard for an asus p4pe? i845? Brian based motherboard and the echo has completly gone away along Brian with aggressive suppressor option in the makefile. I hope this Brian helps others. I'm curious if you tried experimenting with any settings in the via kt4vl bios? Or did you just swap motherboards first? For bios settings, I mean things like timing settings or other PCI bus attributes, even memory speeds. It's a very good thing to know this motherboard could cause echo problems, so thanks for tracking that down and posting here. But it would be really nice if you could use this motherboard to setup another lab machine to experiment on (i.e. try different bios settings) and see what else you can find... you might be able to really make a breakthrough in helping us all with our echo problems. Thanks, -dana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I give up!!
I have multiple installs, and while I would say I have experienced some of the same problems, thankfully our results were'nt close to yours. Perhaps the real problem was you did not do any lab work, we tested about 6 months before we did our first install. One thing we did wrong was started changing too many variables too quickly, i.e. we used the tdm400p prototype boards instead of equipment we had labbed it all on. If you lab it all at your house first and really beat the crap out of it while you are labbing it you will really improve your chances of success. Second, do not change anything or deviate from the agreed upon proposal regardless of how much pressue the customers puts you under (as long as you properly set expectations up front). After the install and you have hit all of the goals you and your customer agreed on (and all of those goals have been successfully labbed at your house before you agreed to them) bill them, then start adding features and making changed on an hourly rate. The first install we had went pretty badly (we were using the prototype versions of the wcfxs boards). The grandstream phones are also really really crappy phones (echo is going to be damned difficult to get rid of on those). I have only had asterisk crash on me a handful of times and I think we have narrowed that down to leaving the console with debug on IAX and Zap up remotely via ssh over a 24 hour period. I hope you do try again, we have a growing customer base and once our clients started to see the power of asterisk over their existing systems (we still get the rare and strange problem) we usually get them to be pretty loyal. Besides it is people like us who are going to start getting rid of the extremly f-ed up Bell System. Were on a mission, not just to provide cool phone systems but to make voice communications monopoly free! - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 16, 2003 10:04 AM Subject: Re: [Asterisk-Users] I give up!! costas wrote: This is a traumatic story. Maybe you can help the rest of us who are making business decisions using *. Will you or client be looking at any other SIP alternatives? Do you think any problems were with the phone sets themselves? I would say a couple of issues were related to the phones, eg the consultative transfer.. Again, sorry to hear of your troubles. -- Original Message -- From: Dave Alan Caruana [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Thu, 16 Oct 2003 15:21:02 +0200 i've just lost $2000 dollars or so on my first commercial asterisk installation .. i'm running a PIV class server, three Digium Wildcard FXO cards, and 10 Grandstream Budgettone SIP phones. The system was to be a PBX for a small company. After over 2 months of pissing about, the client has had his fill of asterisk problems, and asked me to take my equipment out of the building. Obviously, I haven't been paid for anything. The problems I faced were the following : - initially a problem with asterisk crashing totally when there wasn't an extension to ring .. though this was fixed in a subsequent CVS, it was causing downtime. the client has no unix knowledge, and a script I put in to kick in the asterisk when it shut itself down didn't seem to always work. it also reduced the quality of my subsequent callout requests to something on the lines of the phone server is crashed again regardless of what the problem was - a dialplan problem, where one phone was ringing 10 seconds after the others, at the client's request and they were hearing other phones ring and picking up a non-ringing phone (ok, I can't really blame that on asterisk ..) - echo on the lines .. that after much fiddling around with configurations went from terrible to borderline acceptable. To people not used to digital telephony and computer stuff, the echo was VERY annoying. They used to avoid the phones because they said people would not understand them. - no consultative transfer. The closest I got was to park the call, call the other party, tell him a voce which line the call is parked on and then get him to pick up the call. This is, in my opinion, a very basic feature that is missing on asterisk. The park/ pick up sequence proved too difficult for the clients' secretaries to grasp. - I could not get G729 working properly (license paid up, G729 up and running). In the absence of a manual, the fault solving process was something like ask a question on the mailing list, get a few answers, go to the client, try it out, fail, go back home, send another question on the mailinglist with about 48 hours for each iteration. I was also appearing a real chimp expermimenting stuff at the clients' office. At this point I decided to cut my losses, retreive the equipment and call it a day. When asterisk is well documented
Re: [Asterisk-Users] WCFXO echo rexolved for me
Agressive Suppressor is active in the makefile. - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 15, 2003 8:07 AM Subject: Re: [Asterisk-Users] WCFXO echo rexolved for me I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Are you saying you have the aggressive suppressor active or inactive now? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WCFXO echo rexolved for me
Hello, I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH 43017 Phone: 614-798-9106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP X100P Echo Problems
Like most others on this list I also have some really annoying echo whenever a call goes out to the PSTN from a SIP phone... SNOM/Budgettone - Asterisk - X100P - PSTN I have tried every echo canceler in the makefile and turned on and off aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and I can get it reduced to only a few seconds on the intro of the call and after silence, as well as a really annoying beep every so often, and some audio artifacts. I am using KT400 VIA based motheroards MSI KT4VL and x100P and the new tdm400p. Calls within the pbx sound great something to really be proud of (congrats to all of you developers), but going out to the PSTN is extremely annoying. Who do I pay and how much to get rid of this extrordinarily annoying echo from sip - pstn calls? I am not kidding and I also hope others on this list who are making money on asterisk would chip in to help out. This problem is killing me. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH 43017 Phone: 614-798-9106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
