[Asterisk-Users] error in asterisk and LOTS OF log files generated
Our Asterisk server went NUTS sometimes, and when it happens, it will generate log files in seconds almost fill out the /var/log/asterisk directory, following is the console output: Asterisk Event Logger restarted Rotated Logs Per SIGXFSZ (Exceeded file size limit) Apr 20 13:35:05 WARNING[19797]: format_wav.c:491 wav_write: Bad write (0): File too large == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Rotated Logs Per SIGXFSZ (Exceeded file size limit) Apr 20 13:35:05 WARNING[19797]: file.c:233 ast_writestream: Translated frame write failed == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Rotated Logs Per SIGXFSZ (Exceeded file size limit) Apr 20 13:35:05 WARNING[19797]: format_wav.c:491 wav_write: Bad write (0): File too large == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Rotated Logs Per SIGXFSZ (Exceeded file size limit) Anyone know what happened? Thanks in advance. BZ William Zhang ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Building Asterisk with CentOS
Hi, Anyone built Asterisk in CentOS? Any luck on Zaptel module like wcfxo? I just could not build it. Thanks in advance. Bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata.conf setup for TE410P
Hi, I have a TE410P with 3 E1 being enabled, some how it crashes for 2 times lately, I suspect it might be the channel setup issue, can anyone tell me if following part in zapata.conf is correct? switchtype = euroisdn signalling = pri_cpe pridialplan=local group = 1 context = incoming channel => 1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 Also, is there way to log the reason why Asterisk is crashed? Thank you. B ___ 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] Billing software for Asterisk?
Anyone knows there exist such a software that is working with Asterisk? Thanks = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer (again!)
GS phone does blind transfer only. Afer pressing transfer, you will hear dialtone and then dial the number, after dial the whole number, either wait more than 5 seconds or press "redial/send" button, then hangup, it should work. --- "WipeOut ." <[EMAIL PROTECTED]> wrote: > These are probably more issues for grandstream.. Maybe mail > [EMAIL PROTECTED] with the issues about dropping both calls > when the phone is hung up.. > > Later > > > Hello, > > > > I am building an asterisk PBX with some stuff to make a usable VOIP > / > > PSTN Gateway. I use the following devices: > > SIP Phones from GrandStream for VOIP side > > OpenLine4 from voicetronix for PSTN Side > > > > I am building things step by step with some priorities. > > > > I have now a working system able to place and receive calls from/to > pstn. > > > > Before attempting to bring other functions (like voice messaging) > up i > > want to have a proper call transfert functionnality. > > I can't have either blind transfert or consultative transfert > working > > properly. I am VERY interested in consultative transfert but I > don't > > see where and how 'transfer', 'flash' or 'hold' keys and handle in > > asterisk code. > > > > What I would like to do is: > > A and B are taking each other > > A press flash key: B listens music (thet works) and A can call C > > A and C can talk each other but there is no mean for A to transfert > B to > > C. Where should I patch the code to be able to do that? > > Here A can talk either with B or C by pressing on 'Flash' Key but > can't > > hang up any call. > > > > > > IF C is Unavalaible, I haven't seen how to get B back > > > > I welcome any idea about transfert application as it is a main > issue for me: > > AGI application, > > Use of Transfert built in > > Proper use of extension.conf file, > > Patch to the source code of asterisk (I am able to do such a patch > but I > > don't know where to look... chan_sip? apps directory, other?) > > > > Best ragards, > > > > Daniel > > > > -- > > Daniel ANDRE (mailto:[EMAIL PROTECTED]) > > IRIS Technologies - http://www.iris-tech.com > > Serveur kwartz - http://www.kwartz.com > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > __ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID not working
Are those caller ID numeric or have some alpha characters? GS LCD can display only some of those characters. --- John Brown <[EMAIL PROTECTED]> wrote: > I have the following: > > Call -> PSTN -> * -> GrandStream 101 1.0.3.81 > > The GS displays "ohn ro n2600" when the call > is past to the GS. > > If I pass the call to a XTEN client, Caller ID > shows up. > > > Any ideas ?? > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone Defective Units
Greg, The Network Router MAC address issue is resolved by new firmware 1.0.3.81. --- Greg Renouf <[EMAIL PROTECTED]> wrote: > Out of the 8 test Grandstreams we purchased in May- 3 of them are now > totally dead. It looks like power supply problems... > > 2 of the phones have broken buttons... > > And every time I plug them into the network- half of them cause the > network > to crash. It seems that in the network we are testing on (a Dutch > university) does not agree with the Grandstream phones. When plugged > into > the network, they seem to grab the MAC address from the local router > and us > it is their own. This produces total chaos- local pc's get confused > and the > network stops working. Not good. > > I told GS about this problem a few months ago- they initially blamed > the > problem on the network. So far, there has been no satisfactory > solution. > > It seems obvious to me that the GS is not ready for prime time... > > -GSR > > > - Original Message - > From: "Steve Haehnichen" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Saturday, August 23, 2003 5:18 AM > Subject: Re: [Asterisk-Users] Grandstream Budgetone Defective Units > > > > -=> On Fri, 22 Aug 2003 06:49:53 +, "WipeOut ." > <[EMAIL PROTECTED]> said: > > > > > The remote NTP problem is fixed in the current beta firmware that > I > > > have been testing for GS.. Hopefully it will be released soon.. > > > > I'm going 'round and 'round with the GS support guys. I've tried > > three beta versions of the firmware and still don't have a fix for > > what appears to be a serious problem. > > > > If I use a DNS server inside my LAN (within the subnet) while my > > router is not at that same address, the phones crash at boot time. > In > > other cases, they will ARP crazy things, like remote servers or > > 0.0.0.0. I could be wrong about the specifics, but there seems to > be > > certain sets of network addresses that they handle poorly. > > > > I've sent them a number of packet traces and some repeatable > > scenarios, but each firmware update seems to fix one and break > others. > > At this point, I don't have a lot of confidence in their TCP/IP > stack. > > > > That said, none of the three units (or their power supplies) have > > failed on me. :) > > > > For best results, make sure the DHCP server and Router are the same > > machine, and that the DNS server is outside your subnet. > > > > -Steve > > ___ > > 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 = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New budgetone firmware
There is a release note now. --- Paul Barrett <[EMAIL PROTECTED]> wrote: > Hi > > I upgraded earlier today and so far have found that if the Daylight > Saving > option is on one hour is added to the time received from the NTP > server > regardless of date. > > This is using my internal NTP server but I can't get it to work with > any > external NTP server, it simply does not download the date > > Other than this I have seen no change since I upgraded. > > Although there appears to be no Release notes with this release > > Regards > Paul > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of > Brancaleoni > Matteo > Sent: 14 July 2003 20:42 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] New budgetone firmware > > > Hi. > Has anyone experienced with the new firmware .77 ? > There's Day Light Saving time now, but haven't > time to play with it, till now. > > Matteo. > > -- > Matteo Brancaleoni > Espia System Administrator - IT services > Website : http://www.espia.it > Email : [EMAIL PROTECTED] > > > > ___ > 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 = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bugetone SIP Transfer
Transfer should work for Asterisk. You might want to contact Grandstream Tech support for details. --- "WipeOut ." <[EMAIL PROTECTED]> wrote: > Hi, > > I am running firmware 1.0.3.72 and I am having a problem with SIP > transfers on a bugetone phone.. > > Transfer procedure (according to manual).. > > 1. While in call press the "Transfer" button. > 2. When you hear dialtone enter the number to transfer to and hang up > the phone. > > All this seems to do is hangup the call.. The transfer fails.. > > Thanks for any help.. > > -- > __ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BudgeTone 100 Calling Problems
The new firmware should fix the Inband, OutofBand(2833) and add a new DTMF mode--INFO, it will give user choice of their own to setup DTMF mode for their GS phones. The release will be out very soon. --- Stefano Finetti <[EMAIL PROTECTED]> wrote: > > I'm using happily this cheap phones, but I still have a little > problem. > > Configuring the phone is extremely easy on * and I've a couple of > them > perfectly working, except when i try to call some toll-free number > (in italy > 800xxx ). > > If the number called is an IVR system, often with GrandStream (but > also with > Cisco 7905.h323) it's impossible to make the menu choices via the > Dialpad. > > I think that the inband-dtmf Grandstream issue may cause this > problem. > > Going in the details: > > I call the 803121 (Telecom Italia toll-free number for business > customers). > > I hear indefinitely the ring. They have an IVR. > > I call the 80028 (a toll free number of a popular radio station > in > italy): > > I can perfectly hear the answer of the recorded voice - that is NOT > an IVR - > and then I can talk with the operator. > > Anyone has ideas on how to bypass the problem of the IVR answer not > understood by *? > > -- > Stefano Finetti > Technical Coordinator > Lynx Automotive srl > [EMAIL PROTECTED] > Tel: 199 79 79 30 > Fax: 06 233 227 934 > Linux Registered User #271978 > > _______ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF with grandstream phones
You can contact Grandstream for new firmware release, it fix the DTMF problem, it inband and outofband. --- Michael Bielicki <[EMAIL PROTECTED]> wrote: > I am using a grandstream phone with g729 and alaw odecs and in both > modes I > cannot seem to pass dtmf's, neither inband nor out of band, neither > wthrough > a lcoal server nor through a natted connection. Am I missing > something ? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone Supervised Transfer
You are right, Budgetone 100 series currently do have consulted/supervised transfer. It will have in the future. --- Matteo Brancaleoni <[EMAIL PROTECTED]> wrote: > Hi. > > I was wondering if there's a way to do supervised transfers on > the budgetone 102 . Blind works ok, but can't do supervised. > I thought that with the flash button that could be possible, but > seems I'm wrong > > matteo. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
The latest release 1.0.3.60 will allow you to setup a outbound proxy for the phone and the SIP URI's domain can be set in SIP SERVER field. Any sip messages from the phone will use SIP SERVER field's content as domain for phone's SIP URI. --- Juha Heinanen <[EMAIL PROTECTED]> wrote: > Dan Fernandez writes: > > > In the phone, if I set the outbound proxy to the linksys it > doesn磘 do > > anything. > > i have noticed this too. outbound proxy feature is broken in it. > also, > it doesn't do srv lookups, which would allow leaving outbound proxy > empty. it looks to me that the gs guys still have ways to go before > the > phone is ready for prime time. > > -- juha > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgettone 100 phone Configuration
If your proxy server has the same DNS domain name as your SIP URI, then leave outbound proxy blank. Then, check "yes" for NAT Traversal. Type in the STUN server ip, GS has a STUN server 67.153.142.67, or you can use vovida's 66.7.238.210. BTW, there is a new release, check the web site: www.grandstream.com/y-service.htm. You can configure your GS phone with GS tftp server: 4.3.153.56, the reboot the phone. Since you use 1.0.3.53, you will lose you phone configuration this time, you will not lose any confituration from 1.0.3.58 and later release. --- Dan Fernandez <[EMAIL PROTECTED]> wrote: > Will look into this once someone can help me with the configuration > behind > NAT (without NAT I have no problem) > I am using v1.0.3.53 and a linksys router (the phone IP is > 192.168.1.2) > > I磛e tried in my sip.conf with and without NAT=1. > > In the phone, if I set the outbound proxy to the linksys it doesn磘 > do > anything. If I leave outbound proxy empty it registers and I can > place calls > but no audio either way. I have also tried setting the phone for NAT > and no > NAT (no STUN server). > > Don磘 know what else to try. Can someone please help me? > > > - Original Message - > From: "Greg Renouf" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, June 05, 2003 8:16 PM > Subject: RE: [Asterisk-Users] Budgettone 100 phone Configuration > > > > I'm using v.1.0.3.58 and am experiencing that my phone crashes > every > > time the call reaches about 45 minutes in length. > > > > Has anybody had a similar experience? > > > > -GSR > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Stephen > R. > > Besch > > Sent: 05 June 2003 19:03 > > To: [EMAIL PROTECTED] > > Subject: re: [Asterisk-Users] Budgettone 100 phone Configuration > > > > The updated Budgetone firmware (1.0.3.60) has indeed fixed the > "silent > > DTMF" issue. > > > > >By the way, Grandstream just got the "silent DTMF" problem fixed > for > > me > > >and sent me an updated revision this morning (1.0.3.60). I am > just > > >about to install it, but it may require that I debug my tftp > server, > > >which I haven't tested yet. I'll post the list as soon as I get > the > > >new version loaded. > > -- > > Stephen R. Besch, Ph.D. > > SachsLab > > 320 Cary Hall > > SUNY at Buffalo > > Buffalo, NY 14214 > > (716) 829-3289 x106 > > > > ___ > > 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 mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = William Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anybody know why this message shows on console? This is regarding SIP
NOTICE[40966]: File chan_sip.c, Line 1559 (copy_header): No field 'Record-Route' present to copy NOTICE[40966]: File chan_sip.c, Line 1559 (copy_header): No field 'Record-Route' present to copy Thx. Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
