Re: [asterisk-users] Echo() app doesn't work
--- Tzafrir Cohen [EMAIL PROTECTED] wrote: -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' On which platform is that? Echo is executed, and exists without an error. Tzafrir, thank you very much for responding! Logs look the same everywhere (on all 32-bit platforms where Echo() doesn't work) and on the 64-bit xubuntu (where it does). The log says it exited non-zero, which does not seem normal to me, but nevertheless the log has that on the only working setup already mentioned. I guess it is not the platform but maybe some kernel stuff that breaks the thing... Please anyone, any hint? Thanks in advance! I paste here my original message for reference (no broken lines this time): -Original Message-- Date:Fri, 1 Feb 2008 17:01:56 -0800 (PST) From: Yassen Damyanov [EMAIL PROTECTED] Add to Address BookAdd to Address Book Add Mobile Alert Subject: Echo() app doesn't work To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed. Details: A. Platforms: -- AsteriskNOW 0.6 beta 32bit, updated; -- Debian Etch 32bit with stock kernel and native debian-packaged asterisk 1.2 -- Debian Etch 32bit with custom compiled kernel (Timer frequency 1000 Hz and couple more tweaks) and latest stable asterisk (1.4.17) compiled from source -- Ubuntiu 7.10 Server with native ubuntu packages (1.4.10) -- xUbuntu 7.10 Desktop on x86_64 (=amd64) with native ubuntu packages (1.4.10) Echo() works only on the 64-bit setup. Does not work for all other cases. The Playback() app works fine in *all* cases. (The microphone is tested and works fine, so it's not that simple!) For some of the setups I established two separate extensions and they could talk to each other (so important things work, yes). The logs show the same, that is, just what would be normal: -cut here--- Asterisk Ready. *CLI -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569 -- Accepting UNAUTHENTICATED call from 192.168.2.3: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing [EMAIL PROTECTED]:1] Verbose(IAX2/yassen-2, 1|Echo test application) in new stack Echo test application -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' -cut here--- My extentions.conf: -cut here--- [globals] [general] [default] exten = s,1,Verbose(1|Unrouted call handler) exten = s,n,Answer() exten = s,n,Wait(1) exten = s,n,Playback(tt-weasels) exten = s,n,Hangup() [outgoing_calls] [incoming_calls] [internal] exten = 500,1,Verbose(1|Echo test application) exten = 500,n,Echo() exten = 500,n,Hangup() exten = 501,1,Verbose(1|Playback test application) exten = 501,n,Playback(vm-review) exten = 501,n,Wait(1) exten = 501,n,Hangup() [phones] include = internal -cut here--- My iax.conf: -cut here--- [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no autokill=yes [yassen] type=friend host=dynamic context=phones -cut here--- Anyone having a suggestion what might be the reason for the nonworking Echo() ? I am really stuck; Google could not really help. Any ideas would be highly appreciated! Thanks in advance, Yassen -- Yassen Damyanov Adelie Ltd. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards
Ron Joffe ha scritto: On Friday 01 February 2008 15:31, Matt wrote: It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone know if there are plans for a PCI-e analog card for FXO use? I have been using 220B's for about 6 months. I have about 20 of them out in the field. I have not had any issues with them, and feedback is positive. Same here. I've been using five TE220B in my company at 5 different sites since october 2007; up to now, zero problems and no echo at all. One of the sites runs a small callcenter that handles about 1000 incoming calls per day. So far the feedback is really positive. Alberto. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA with pulse dialing support over FXS
Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards from the plastic case and to fit them into the phones after making the appropriate changes to the phones' exterior to add holes for rj-45 socks and dc power input) Thanks. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel timer on Intel Dual Core servers
Friends, I'm having severe problems with zaptel timers on Intel Dual Core systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM or PRI cards - all ends up with large timer probems - zttest going down to 50% accuracy on some systems, even to -1 on ztdummy systems and voice quality is no more. A restart is the only way to get back to a working system. We're only running a few meetme's with a total of 10 participants, so the load is not excessive. Only SIP calls, nothing else (it's me, right :-) ) ? If I hit the system with SIPP targeting an extension only doing playback, which doesn't even use zttimer, the timer slides away. I've played around with the irqbalance daemon, but don't see any big differences with or without it. My question is if anyone else have seen this and if anyone has a possible solution? Thanks in advance for your help! /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone support SIP and IAX
On Jan 22, 2008 3:40 PM, Jared Smith [EMAIL PROTECTED] wrote: I received a couple of ALL7960 phones from ALLNET Network in Germany this past week, and the firmware handles both SIP and IAX. I haven't done a lot of stress testing on them, but so far I'm very happy with the phones and their ability to easily make both SIP and IAX calls. Jared et al, I have had one of these phones for about a week, with only a little time to play with it. I wrote a very incomplete review that I hope to finish RSN but here's a few observations: http://voipusersconference.org/review and I do hope to add more meat to it. Michael Graves has posted a bunch of photos on that site as well. My very first evaluation is that this would be a perfect first phone for an Asterisk newbie, replacing for example the Grandstream BT100 which is inferior in many ways but also cheaper than the Allnet. The Allnet can be set to do SIP or IAX2 in the web interface, but the unique feature AFAIK is that it can be set to do both so you can dial a (configurable) *0 for SIP and *1 for IAX. This is great for testing. The phone can register with up to four servers and it can speed dial any number. That means it can speed dial a SIP URI and I haven't figured out if this is doable on either the Polycom ip500 or the Sipura SPA-941 that I own. The voice quality seems very good to me and the phone has a much sturdier feel than most of the competition in this price range. I'm told the US dealer, somewhere in Florida, hasn't answered the call put in a few days ago yet. In the meantime, there is a UK dealer http://www.allnetuk.com/ The dealer I was told to contact by the German wholesaler to buy the phone in France is called Senso Telecom: http://www.senso-telecom.com/content/view/13/28/lang,en/ Here's the wholesaler page with other countries on it: http://212.18.29.54 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timer on Intel Dual Core servers
My question is if anyone else have seen this and if anyone has a possible solution? Nearly all of the boxes we've built over the last few months have been either Core2 Duo or Core2 Quad based and we haven't seen any timer issues. In most cases these are purely IP boxes (no PRI or TDM cards) with ztdummy. We've had plenty of meetme conferences over that time and I've not noticed any problems. I'll run zttest on a couple of them over the weekend and see what results we get, but certainly, there haven't been any complaints about quality during conferences. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timer on Intel Dual Core servers
Likewise here. The company I work for sells duo core boxes (though mostly /with/ E1 cards) and we have no issue with timing. Chris Bagnall wrote: My question is if anyone else have seen this and if anyone has a possible solution? Nearly all of the boxes we've built over the last few months have been either Core2 Duo or Core2 Quad based and we haven't seen any timer issues. In most cases these are purely IP boxes (no PRI or TDM cards) with ztdummy. We've had plenty of meetme conferences over that time and I've not noticed any problems. I'll run zttest on a couple of them over the weekend and see what results we get, but certainly, there haven't been any complaints about quality during conferences. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 - Problems with SIP/REFER
I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below. There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context phones and are set to not allow reinvites. All phones can dial each other directly. The dialplan looks as follows: [phones] Exten = 5253,1,Dial(SIP/5253,10) Exten = 5878,1,Dial(SIP/5878,10) Exten = 101,1,Dial(SIP/[EMAIL PROTECTED],10) Transfer fails regardless of the order (101 calls 5878, 5878 transfers to 5253 or 5878 calls 5253, 5253 transfers to 101, etc) I do not understand the message Spawn Extension (phones, 101, 0) exited non-zero in the debug - there is no priority zero in a dialplan - priority should start at 1. What is this message telling me? What do I need to do to allow these phones to transfer calls between each other? Any help is greatly appreciated! Here is the debug: *CLI == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/5253-0823eab0, SIP/5878) in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 10.7.10.1 port 19968 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.7.10.51:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport Max-Forwards: 70 From: 5253 sip:[EMAIL PROTECTED];tag=as05a48c1a To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-beta2 Remote-Party-ID: 5253 sip:[EMAIL PROTECTED];privacy=off;screen=no Date: Wed, 30 Jan 2008 01:12:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 259 v=0 o=root 864806723 864806723 IN IP4 10.7.10.1 s=Asterisk PBX 1.6.0-beta2 c=IN IP4 10.7.10.1 t=0 0 m=audio 19968 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 5878 --- SIP read from UDP://10.7.10.51:5060 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport=5060;received=10.7.10.1 From: 5253 sip:[EMAIL PROTECTED];tag=as05a48c1a To: sip:[EMAIL PROTECTED]:5060;tag=694417843 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: sip:10.7.10.1;appearance-index=1 Contact: 5878 sip:[EMAIL PROTECTED]:5060 Server: Aastra 53i/2.1.0.2145 Content-Length: 0 - --- (12 headers 0 lines) --- -- SIP/5878-08250098 is ringing --- SIP read from UDP://10.7.10.51:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport=5060;received=10.7.10.1 From: 5253 sip:[EMAIL PROTECTED];tag=as05a48c1a To: sip:[EMAIL PROTECTED]:5060;tag=694417843 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: sip:10.7.10.1;appearance-index=1 Contact: 5878 sip:[EMAIL PROTECTED]:5060 Server: Aastra 53i/2.1.0.2145 Supported: timer, replaces Content-Type: application/sdp Content-Length: 313 v=0 o=MxSIP 0 0 IN IP4 10.7.10.51 s=SIP Call c=IN IP4 10.7.10.51 t=0 0 m=audio 3000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ZXJ1UmhNLDFmQGNHYGAnRlpKbjEudk9Gfjh8blo/ a=fmtp:101 0-15 a=ptime:20 a=sendrecv - --- (14 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.7.10.51:3000 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.7.10.51:3000 list_route: hop: sip:[EMAIL PROTECTED]:5060 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 10.7.10.51, port 5060 Transmitting (NAT) to 10.7.10.51:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK6476d991;rport Max-Forwards: 70 From: 5253 sip:[EMAIL PROTECTED];tag=as05a48c1a To: sip:[EMAIL PROTECTED]:5060;tag=694417843 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0-beta2 Remote-Party-ID: 5253 sip:[EMAIL PROTECTED];privacy=off;screen=no Content-Length: 0 --- -- SIP/5878-08250098 answered SIP/5253-0823eab0 -- Packet2Packet bridging SIP/5253-0823eab0 and
Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards
Same here. I've been using five TE220B in my company at 5 different sites since october 2007; up to now, zero problems and no echo at all. One of the sites runs a small callcenter that handles about 1000 incoming calls per day. So far the feedback is really positive. Great! Thanks guys. This makes me feel good about purchasing them. I never had any issues with Digium's cards when in the PCI line (other than the whole BIOS+IRQ issue... which forced me to go to Sangoma PCI-e cards). Now that Digium has PCI-e cards, I'm anxious to switch back. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA with pulse dialing support over FXS
On Saturday 02 February 2008 02:45:09 Alberto Pastore wrote: Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards from the plastic case and to fit them into the phones after making the appropriate changes to the phones' exterior to add holes for rj-45 socks and dc power input) As an aside, note that you'll void your warranty if you remove the IAXy from its case, but more importantly, please remember to also add ventilation holes to the phone, near where you place the circuitry from the IAXy. The venting of heat is perhaps one of the more important functions of the IAXy case (and one that if you don't replicate, you'll regret later). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA with pulse dialing support over FXS
On Saturday 02 February 2008 02:45:09 am Alberto Pastore wrote: I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards from the plastic case and to fit them into the phones after making the appropriate changes to the phones' exterior to add holes for rj-45 socks and dc power input) look into Rotatone: http://www.oldphoneworks.com/antique-phone-parts/by-type/part.asp?currency=USDPhonePart=729PhonePartType=138 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Matthew J. Roth a écrit : Administrator TOOTAI wrote: This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards Daniel, I thought that using an empty TDM400P as a timing source may no longer be the best solution due to the emergence of new stable timing sources (such as HPET), but this is an interesting issue. Are you stating that you can't put an X100P or a TDM400P with no lines attached alongside a B410P because it impacts the stability of Asterisk? Yes Do you have any idea why? No Can't the B410P be used as a timing source? No What have you done to provide stable timing? ztdummy, not always stable :-( I know that's a lot of questions, but I'm genuinely curious. ;-) It seems very strange that a TDM400P in timingonly mode and no lines attached would have any impact on Asterisk's stability. I have to add that this is mainly true with 2 B410P in the server or with B410P in amd64 machines. Seems also that Debian Etch with a 2.6.18 kernel is not the best :-( -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Softphones and Citrix ?
d4rk f1br a écrit : Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I MozPhone (moziax.mozdev.org) has been designed to be run in thin client environment. I don't know much about Citrix, but I have customers running MozPhone on thin clients in a Terminal Server environment. The idea is that sound and IAX communication are managed locally on the thin client, while user interface (Firefox extension) runs on the terminal server. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timer on Intel Dual Core servers
2 feb 2008 kl. 14.10 skrev Rob Hillis: Likewise here. The company I work for sells duo core boxes (though mostly with E1 cards) and we have no issue with timing. Chris Bagnall wrote: My question is if anyone else have seen this and if anyone has a possible solution? Nearly all of the boxes we've built over the last few months have been either Core2 Duo or Core2 Quad based and we haven't seen any timer issues. In most cases these are purely IP boxes (no PRI or TDM cards) with ztdummy. We've had plenty of meetme conferences over that time and I've not noticed any problems. I'll run zttest on a couple of them over the weekend and see what results we get, but certainly, there haven't been any complaints about quality during conferences. Thanks for the feedback. Makes me even more worried about these servers. Might have to check the Linux installation to make sure they have installed all the proper drivers for the dual core processors. Something's very wrong. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Softphones and Citrix ?
On Feb 2, 2008 12:09 PM, Jean-Denis Girard [EMAIL PROTECTED] wrote: d4rk f1br a écrit : Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I MozPhone (moziax.mozdev.org) has been designed to be run in thin client environment. I don't know much about Citrix, but I have customers running MozPhone on thin clients in a Terminal Server environment. The idea is that sound and IAX communication are managed locally on the thin client, while user interface (Firefox extension) runs on the terminal server. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 Awesome! Thanks for pointing out this link. I was looking for a thinclient softphone a couple of years ago. Any feedback on how well this works (thinclient and/or PC browser plugin)? Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parking Valet
Did you ever get this to work? I can get it to park calls, but it will not announce what parking space all in the same cycle, I can't see how this is useful is you have to transfer to park it, then dial another extension just to see where it was parked! Anyone got this working? On Jul 18, 2007 6:58 PM, Russell Bryant [EMAIL PROTECTED] wrote: Kevin Kiely wrote: app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/ Indicates support for Asterisk 1.4. The documentation listed suggests an install like so: cd /usr/src/asterisk cp contrib/scripts/astxs /usr/bin/ cd apps wget http://www.bkw.org/app_valetparking.c cd .. astxs -install apps/app_valetparking.c However astxs doesn't seem to be present in asterisk 1.4 Does anyone have this working with 1.4? and any suggestions on how to install? astxs won't work with the build system in 1.4, so it's not there anymore. However, if you drop the file into the apps directory, Asterisk will automatically build and install it for you when you run make and make install. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
Actively maintained or actively being broken and fixed with constant updates? Not something suitable for Production IMHO. Makes more sense for development and experimentation IMHO. -Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Friday, February 01, 2008 1:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Enterprise or Fedora? shadowym [EMAIL PROTECTED] writes: I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. The advantage of Fedora is that it is very actively maintained -- and asterisk is only a yum install asterisk away! /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timer on Intel Dual Core servers
I think CentOS 5 is better with Dual/Quad core than CentOS 4. I have no direct technical evidence of this. Just empirical from the Google oracle. From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Saturday, February 02, 2008 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel timer on Intel Dual Core servers Likewise here. The company I work for sells duo core boxes (though mostly with E1 cards) and we have no issue with timing. Chris Bagnall wrote: My question is if anyone else have seen this and if anyone has a possible solution? Nearly all of the boxes we've built over the last few months have been either Core2 Duo or Core2 Quad based and we haven't seen any timer issues. In most cases these are purely IP boxes (no PRI or TDM cards) with ztdummy. We've had plenty of meetme conferences over that time and I've not noticed any problems. I'll run zttest on a couple of them over the weekend and see what results we get, but certainly, there haven't been any complaints about quality during conferences. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Real API for Perl?
Alex Balashov wrote: Ken D'Ambrosio wrote: Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI binding improved since then? 2) Is there any chance of a real API for Perl? What is your criterion of real? That is to say, what do you need that it does not provide? I've used AGI and FastAGI in Perl extensively and it is yet to fail to serve my purposes. Maybe Ken is referring to a pre made framework like Asterisk-Java or PasAGI. I don't know perl so maybe there *is* a framework already. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Softphones and Citrix ?
Steve Totaro a écrit : Awesome! Thanks for pointing out this link. I was looking for a thinclient softphone a couple of years ago. Any feedback on how well this works (thinclient and/or PC browser plugin)? Disclaimer: as the main developper of MozPhone I'm obviously biased :) I used it myself in a Linux LTSP environment years ago without trouble. My customers using it in the windows TS environment did lot of testing. The audio part of the thin client is obviously important. They use Neoware thin clients, and were getting good results with onboard sound card. Then they tried a first usb headset which gave very bad sound, and finally settled on the Plantronics CS60-usb headset. They contracted me to add managment of CS60 buttons (off/on hook, mute) to MozPhone, so I guess they are satisfied. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Real API for Perl?
Well, no, there really aren't any prebuilt high-level frameworks for approaching Asterisk through the Manager API or AGI. Instead, there are just AGI bindings that allow you to integrate dial plan logic with outboard code. I always figured that was kind of the whole point of such bindings, so nothing about it strikes me as incomplete or lacking in a sufficient degree of reality. The only difference between this and Asterisk-java is simply that the latter encapsulates many of these actions in more high-level wrappers, which is likely to be a concession to the phenomenology of Java Thinking(TM) more than anything else. Your mileage may vary. Lee Jenkins wrote: Alex Balashov wrote: Ken D'Ambrosio wrote: Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI binding improved since then? 2) Is there any chance of a real API for Perl? What is your criterion of real? That is to say, what do you need that it does not provide? I've used AGI and FastAGI in Perl extensively and it is yet to fail to serve my purposes. Maybe Ken is referring to a pre made framework like Asterisk-Java or PasAGI. I don't know perl so maybe there *is* a framework already. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timer on Intel Dual Core servers
Johansson Olle E [EMAIL PROTECTED] writes: Friends, I'm having severe problems with zaptel timers on Intel Dual Core systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM or PRI cards - all ends up with large timer probems - zttest going down to 50% accuracy on some systems, even to -1 on ztdummy systems and voice quality is no more. A restart is the only way to get back to a working system. It's a bit difficult to guess what is wrong. You may want to look at which kernel version you use, and the options used. Perhaps trying maxcpus=1 on the SMP kernel would be interesting. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_valetparking.c anyone using it on 1.4?
Hi List, I have this running, but after I park a call it will not announce where it is at, it's like you have to call another application just to say where it is parked at. I have tried a second priority option for the same extension with that ValetParkList but it seems once ValetParkCall has been ended it will not process anymore priorities in this extension. Any ideals or help would be great! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple SIP phones behind a Linksys firewall
I posted an email a few days regarding a problem with hearing the voicemail greeting on my sip phones. It turns out to be a phone/stun/linksys issue - not an asterisk issue. Which brings up a couple of questions I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. But apparently thats not the case. Is it a Linksys bug, a Grandstream bug in the BudgeTone-100 phone, or am I off base and just doing something wrong? I cleary have problems as soon as I try to use a second phone behind the Linksys - registration issues, cant hear voicemail greeting, etc.,. My next test was to run multiple STUN servers on the same machine with different ports. Then, for my multiple SIP phones behind the Linksys, have each phone use a different stun port. Any thoughts? John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timer on Intel Dual Core servers
Disabling ACPI might provide some interesting results as well. Core Duo systems will drop to one processor. I've seen some very odd timing problems with VMWare products on servers with ACPI turned on in the past. On Feb 2, 2008 2:58 PM, Benny Amorsen [EMAIL PROTECTED] wrote: Johansson Olle E [EMAIL PROTECTED] writes: Friends, I'm having severe problems with zaptel timers on Intel Dual Core systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM or PRI cards - all ends up with large timer probems - zttest going down to 50% accuracy on some systems, even to -1 on ztdummy systems and voice quality is no more. A restart is the only way to get back to a working system. It's a bit difficult to guess what is wrong. You may want to look at which kernel version you use, and the options used. Perhaps trying maxcpus=1 on the SMP kernel would be interesting. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Shared line appearance phones?
Any examples yet Russell? Thanks! On Dec 3, 2007 6:43 PM, shadowym [EMAIL PROTECTED] wrote: That would be VERY much appreciated Russell, There seems to be a lack of info and the accompanying confusion/misinformation about this. -Original Message- From: Russell Bryant [mailto:[EMAIL PROTECTED] Sent: Friday, November 30, 2007 4:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Shared line appearance phones? Mark Wiater wrote: I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID from a PRI on several Polycom IP430's. Might you be willing to share some specific configurations for such a situation? There are some basic examples in doc/sla.pdf in the 1.4 tree. However, I have on my to-do list to spend a week with an SLA test environment and coming up with an extensive set of examples of the different ways it can be used. I will post something to this list when that is available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 - Problems with SIP/REFER
- Original Message From: Jake Wicke [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Friday, 1 February, 2008 5:34:12 PM Subject: [asterisk-users] Asterisk 1.6 - Problems with SIP/REFER I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below. There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context “phones” and are set to not allow reinvites. All phones can dial each other directly. The dialplan looks as follows: [phones] Exten = 5253,1,Dial(SIP/5253,10) Exten = 5878,1,Dial(SIP/5878,10) Exten = 101,1,Dial(SIP/[EMAIL PROTECTED],10) Transfer fails regardless of the order (101 calls 5878, 5878 transfers to 5253 or 5878 calls 5253, 5253 transfers to 101, etc) I do not understand the message “Spawn Extension (phones, 101, 0) exited non-zero” in the debug – there is no “priority zero” in a dialplan – priority should start at 1. What is this message telling me? What do I need to do to allow these phones to transfer calls between each other? Any help is greatly appreciated! Hi Jake, I don't have the answer but I did look at your trace and something about the way the transfer is being done from your phones is not quite right. You're calling 5253, then calling 5878 and then requesting a blind transfer of 5253 to 5878. However at that stage 5878 is already on the phone. I suspect the transfer should be being requested as an attended one not a blind one. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP: IP in the VIA-Header
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I have these settings in my sip.conf: [general] nat=yes canreinvite=no externip = tinloaf.dyndns.org localnet=192.168.0.0/255.255.0.0 (if you need the complete sip.conf i can put in on some webserver) Now I use my asterisk on a NATed box with the IP 192.168.1.100 as an outbound proxy for a SIP-Softphone in the same network (running on 192.168.1.3). Asterisk relays all SIP messages, but I would expect it to include my public IP (tinloaf.dyndns.org) in the VIA-headers. This is how outgoing packets look like (The lines starting with U or # are from ngrep): = U 2008/02/02 22:05:48.730998 192.168.1.100:5060 - 69.90.155.70:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK79892063;rport. From: Lukas Barth sip:[EMAIL PROTECTED];tag=as2c94d5ce. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]:5060. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Sat, 02 Feb 2008 21:05:48 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 264. . v=0. o=root 16173 16173 IN IP4 192.168.1.100. s=session. c=IN IP4 192.168.1.100. t=0 0. m=audio 9120 RTP/AVP 3 0 8 101. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 2008/02/02 22:05:48.912033 69.90.155.70:5060 - 192.168.1.100:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK79892063;rport=5060;received=90.134.116.93. From: Lukas Barth sip:[EMAIL PROTECTED];tag=as2c94d5ce. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Content-Length: 0. . === Now how can i tell asterisk to put my public IP (in this case 90.134.116.93) into the VIA-field? Kind regards, Lukas Barth -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHpNwjgsbFi6ZpoGERAgK1AJ0YJ9cb16QET3lQF+pzfTyELLQOyACdG0rj IvrqJqaGOVtVNT5ZAt4WgW8= =KYTQ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall
On Feb 2, 2008, at 2:11 PM, John Von Essen [EMAIL PROTECTED] wrote: I posted an email a few days regarding a problem with hearing the voicemail greeting on my sip phones. It turns out to be a phone/stun/linksys issue - not an asterisk issue. Which brings up a couple of questions I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. But apparently thats not the case. Is it a Linksys bug, a Grandstream bug in the BudgeTone-100 phone, or am I off base and just doing something wrong? I cleary have problems as soon as I try to use a second phone behind the Linksys - registration issues, cant hear voicemail greeting, etc.,. My next test was to run multiple STUN servers on the same machine with different ports. Then, for my multiple SIP phones behind the Linksys, have each phone use a different stun port. Any thoughts? John I have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 - Problems with SIP/REFER
1 feb 2008 kl. 18.34 skrev Jake Wicke: I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below. When you have issues, it's always a good idea to check the bug tracker. There might be other people having the same issues, in some cases, there's also a solution. If you don't find it on first search, make sure you also search in resolved and closed issues. For this case, we do have open bug reports. The same issue exists in 1.4 and we're working on a solution. Stay tuned. Have a nice weekend! /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall
Greg, Without STUN how are the phones able to register? I was unable to get the Grandstream phones to work at all without STUN. -John From : Greg Oliver [EMAIL PROTECTED] To : Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall Date : Sat, 2 Feb 2008 15:15:34 -0600 On Feb 2, 2008, at 2:11 PM, John Von Essen [EMAIL PROTECTED] wrote: I posted an email a few days regarding a problem with hearing the voicemail greeting on my sip phones. It turns out to be a phone/stun/linksys issue - not an asterisk issue. Which brings up a couple of questions I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. But apparently thats not the case. Is it a Linksys bug, a Grandstream bug in the BudgeTone-100 phone, or am I off base and just doing something wrong? I cleary have problems as soon as I try to use a second phone behind the Linksys - registration issues, cant hear voicemail greeting, etc.,. My next test was to run multiple STUN servers on the same machine with different ports. Then, for my multiple SIP phones behind the Linksys, have each phone use a different stun port. Any thoughts? John I have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Ubuntu has a real time kernel in repository apt-get install linux-rt . So you dont need to recompile . I think debian should also have one in repository , or u can manually compile a real time enabled kernel . Here's what is shows with real time patched kernel . dmesg|grep ztdummy [ 53.293071] ztdummy: Trying to load High Resolution Timer [ 53.293076] ztdummy: Initialized High Resolution Timer [ 53.293078] ztdummy: Starting High Resolution Timer [ 53.293080] ztdummy: High Resolution Timer started, good to go zttest Opened pseudo zap interface, measuring accuracy... 100.00% 99.987793% 99.792480% 99.780273% 99.987793% 99.975586% 99.987793% 99.987793% 100.00% 100.00% 100.00% 99.987793% 99.987793% 99.987793% 100.00% 100.00% 99.987793% 100.00% --- Results after 18 passes --- Best: 100.00 -- Worst: 99.780273 -- Average: 99.969482 On Feb 2, 2008 10:27 PM, Administrator TOOTAI [EMAIL PROTECTED] wrote: Matthew J. Roth a écrit : Administrator TOOTAI wrote: This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards Daniel, I thought that using an empty TDM400P as a timing source may no longer be the best solution due to the emergence of new stable timing sources (such as HPET), but this is an interesting issue. Are you stating that you can't put an X100P or a TDM400P with no lines attached alongside a B410P because it impacts the stability of Asterisk? Yes Do you have any idea why? No Can't the B410P be used as a timing source? No What have you done to provide stable timing? ztdummy, not always stable :-( I know that's a lot of questions, but I'm genuinely curious. ;-) It seems very strange that a TDM400P in timingonly mode and no lines attached would have any impact on Asterisk's stability. I have to add that this is mainly true with 2 B410P in the server or with B410P in amd64 machines. Seems also that Debian Etch with a 2.6.18 kernel is not the best :-( -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timer on Intel Dual Core servers
On Sat, Feb 02, 2008 at 09:46:42AM +0100, Johansson Olle E wrote: Friends, I'm having severe problems with zaptel timers on Intel Dual Core systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM or PRI cards - all ends up with large timer probems - zttest going down to 50% accuracy on some systems, even to -1 on ztdummy systems and voice quality is no more. A restart is the only way to get back to a working system. A restart of asterisk? Not touhcing the kernel modules? What CPU is it, exactly? /proc/cpuinfo would help. What version of Zaptel? Asterisk? Kernel? distribution? What happens if you don't run those meetme-s? We're only running a few meetme's with a total of 10 participants, so the load is not excessive. Only SIP calls, nothing else (it's me, right :-) ) ? If I hit the system with SIPP targeting an extension only doing playback, which doesn't even use zttimer, the timer slides away. When exactly does a playback to a SIP channel use timing from zaptel? I know that bad zaptel timer is known to mess sound playback anywhere. I just don't fuly understand why. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
I prefer CentOS barebone install and yumming the way up for dependencies but manually compile asterisk/zaptel . Ubuntu servers are pretty good too since its repositories are quite bigger compared to CentOS . On Feb 2, 2008 11:45 PM, shadowym [EMAIL PROTECTED] wrote: Actively maintained or actively being broken and fixed with constant updates? Not something suitable for Production IMHO. Makes more sense for development and experimentation IMHO. -Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Friday, February 01, 2008 1:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Enterprise or Fedora? shadowym [EMAIL PROTECTED] writes: I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. The advantage of Fedora is that it is very actively maintained -- and asterisk is only a yum install asterisk away! /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall
On Feb 2, 2008, at 3:43 PM, [EMAIL PROTECTED] wrote: Greg, Without STUN how are the phones able to register? I was unable to get the Grandstream phones to work at all without STUN. -John I have nat on in sip.conf and off on the phones. Works perfect for 7960/1 and 7971. When I get back home, I will login to the asterisk servers and tell you what IPs the registration requests have in them. From : Greg Oliver [EMAIL PROTECTED] To : Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall Date : Sat, 2 Feb 2008 15:15:34 -0600 On Feb 2, 2008, at 2:11 PM, John Von Essen [EMAIL PROTECTED] wrote: I posted an email a few days regarding a problem with hearing the voicemail greeting on my sip phones. It turns out to be a phone/stun/linksys issue - not an asterisk issue. Which brings up a couple of questions I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. But apparently thats not the case. Is it a Linksys bug, a Grandstream bug in the BudgeTone-100 phone, or am I off base and just doing something wrong? I cleary have problems as soon as I try to use a second phone behind the Linksys - registration issues, cant hear voicemail greeting, etc.,. My next test was to run multiple STUN servers on the same machine with different ports. Then, for my multiple SIP phones behind the Linksys, have each phone use a different stun port. Any thoughts? John I have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
Ubuntu server for me please simply, is better... install. then activate universe and multiverse repositories sudo apt-get update sudo apt-get upgrade sudo apt-get build-dep asterisk and then... tar xvfz ./configure make make install ... is very very easy and clean, and IMHO i guess is better SO Ubuntu than any other RHEL based distro... IMHO RedHat sucks xP On Sun, 3 Feb 2008 03:59:47 +0530, Jaswinder Singh [EMAIL PROTECTED] wrote: I prefer CentOS barebone install and yumming the way up for dependencies but manually compile asterisk/zaptel . Ubuntu servers are pretty good too since its repositories are quite bigger compared to CentOS . On Feb 2, 2008 11:45 PM, shadowym [EMAIL PROTECTED] wrote: Actively maintained or actively being broken and fixed with constant updates? Not something suitable for Production IMHO. Makes more sense for development and experimentation IMHO. -Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Friday, February 01, 2008 1:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Enterprise or Fedora? shadowym [EMAIL PROTECTED] writes: I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. The advantage of Fedora is that it is very actively maintained -- and asterisk is only a yum install asterisk away! /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
On Sat, Feb 02, 2008 at 05:13:39PM -0600, [EMAIL PROTECTED] wrote: Ubuntu server for me please simply, is better... install. then activate universe and multiverse repositories sudo apt-get update sudo apt-get upgrade sudo apt-get build-dep asterisk and then... tar xvfz ./configure make make install ... is very very easy and clean, and IMHO i guess is better SO Ubuntu than any other RHEL based distro... Mind you, asterisk is in the ubuntu universe (rather than main) archive. As such, it also has some other dependencies in universe. Specifically (looking at the current Hardy) package: libc-client2007-dev libiksemel-dev libopenh323-dev libpri-dev libradiusclient-ng-dev libtonezone-dev libvpb-dev zaptel-source As a rule, the universe packages there get much less attention, and don't necessarily get regular security updates. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall
And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall? In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio. -Original Message- From: Greg Oliver [EMAIL PROTECTED] Sent: Saturday, February 02, 2008 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall On Feb 2, 2008, at 2:11 PM, John Von Essen [EMAIL PROTECTED] wrote: I posted an email a few days regarding a problem with hearing the voicemail greeting on my sip phones. It turns out to be a phone/stun/linksys issue - not an asterisk issue. Which brings up a couple of questions I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. But apparently thats not the case. Is it a Linksys bug, a Grandstream bug in the BudgeTone-100 phone, or am I off base and just doing something wrong? I cleary have problems as soon as I try to use a second phone behind the Linksys - registration issues, cant hear voicemail greeting, etc.,. My next test was to run multiple STUN servers on the same machine with different ports. Then, for my multiple SIP phones behind the Linksys, have each phone use a different stun port. Any thoughts? John I have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall
I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. I got 10-20 SPA942's behind a OpenWRT router (on WRT54G, WRTSL54GS, ...) at several sites, no STUN, no special configuration, no problems at all. Just as a precaution, I set the SIP port and RTP port range for each phone differently so that it's unique (i.e. Phone 1 SIP port 6001 and RTP 10100-10199, etc.) but that's really just a precaution to help the the Linux' conntrack on the OpenWRT a bit. It's not really needed as the router will resolve port conflicts by rewriting the ports transparently. Bottom line, a few phones behind a well-behaved NAT should work just fine. /Luki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall
The server is at a remote datacenter - no nat, no firewall, pure public IP. The phones are at home offices (i.e. DSL or Cable with Linksys-type firewall/routers). My initial testing was with a single SIP phone at the home office - and everything worked fine. But when I have two SIP phones at the home office, things start behaving badly. I understand the issue of phone-to-phone, where both phones are behind a nat at the home office - but that is not the issue I am having. My main problem is when I have two phones at the home office, the second phone cant register, and/or, you cant here the voicemail greeting when you try to check messages. From : Robert Norton - SophTelecom.com [EMAIL PROTECTED] To : Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall Date : Sat, 2 Feb 2008 18:25:16 -0700 And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall? In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio. -Original Message- From: Greg Oliver [EMAIL PROTECTED] Sent: Saturday, February 02, 2008 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall On Feb 2, 2008, at 2:11 PM, John Von Essen [EMAIL PROTECTED] wrote: I posted an email a few days regarding a problem with hearing the voicemail greeting on my sip phones. It turns out to be a phone/stun/linksys issue - not an asterisk issue. Which brings up a couple of questions I always assumed that you can have multiple SIP phones behind a Linksys firewall/router (WRT54G) all using the same STUN server/port. But apparently thats not the case. Is it a Linksys bug, a Grandstream bug in the BudgeTone-100 phone, or am I off base and just doing something wrong? I cleary have problems as soon as I try to use a second phone behind the Linksys - registration issues, cant hear voicemail greeting, etc.,. My next test was to run multiple STUN servers on the same machine with different ports. Then, for my multiple SIP phones behind the Linksys, have each phone use a different stun port. Any thoughts? John I have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
Tzafrir Cohen wrote: On Sat, Feb 02, 2008 at 05:13:39PM -0600, [EMAIL PROTECTED] wrote: Ubuntu server for me please simply, is better... install. then activate universe and multiverse repositories sudo apt-get update sudo apt-get upgrade sudo apt-get build-dep asterisk and then... tar xvfz ./configure make make install ... is very very easy and clean, and IMHO i guess is better SO Ubuntu than any other RHEL based distro... Mind you, asterisk is in the ubuntu universe (rather than main) archive. As such, it also has some other dependencies in universe. Specifically (looking at the current Hardy) package: libc-client2007-dev libiksemel-dev libopenh323-dev libpri-dev libradiusclient-ng-dev libtonezone-dev libvpb-dev zaptel-source As a rule, the universe packages there get much less attention, and don't necessarily get regular security updates. i think, its matter of knowing what you want, I have been running FC5 with sangoma a104dx card for over a year, i have not experienced any serious stress. goksie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom - Buddy Watch not a choice when adding Speed Dial
Check your sip.cfg for the line: feature.1.name=presence feature.1.enabled=1 I would imagine that you have enabled=0 That was it! Thanks - Thermal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
