[Asterisk-Users] RE: Asterisk crashed so often
From: Unavailable ID [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 4 Mar 2004 16:33:12 -0800 Subject: [Asterisk-Users] Asterisk crashed so often Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0250_01C40206.62A69480 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable FYI that I have experience the same problem many times. The system is = running RH 9.0 with Asterisk CVS-02/21/04. Here is the output from the = console: I am using RH 9.0 too. If I forget to do the: export LD_ASSUME_KERNEL=2.4.1 before asterisk is started then it will handle less than 10 calls before it crashes. Otherwise it seems rockstable in our enviroment with SIP and zaptel TP410P's when it's started Freddi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3com NBX phones
On Thu, Mar 04, 2004 at 04:32:52PM -0500, Clif Jones wrote: I know a little history on the 3com SIP phones... We have about a dozen of them where I work. I'm not familiar with the NBX100 model number but the ones we have are labeled: P/N: 655005001. The first ones didn't support SIP out of the box and had to be upgraded with a new flash image. I can't recall if they came from the factory with H323 or the 3com proprietary IP protocol but the phones look just like the 3com PBX phones you see in small businesses. 3com abandoned the phone after spinning off the SIP division (Commworks?) and determining that the phone hardware just didn't have the resources to continue work on SIP. It is a shame because these phones boot faster than any other IP phone I have seen and have a good speakerphone. The image that we use is pre-RFC3261 but would probably work with Asterisk. I have the same phones here (two of them). Got them directly from 3com in late 2000 I think. They were never distributed in the channel I think (at least not here in Europe). Mine arrived SIP ready, and I flashed them once with a standard TFTP procedure. Unfortunatly, 3com discontinued them, no more firmware are available, AFAIK. My firmware is 1.0.1.21.0 SIP. Unfortunatly I'm not able to find this latest firmware on file. Note that the hardware is probably not the same as the standard NBX phones : my SIP phones did feature an IR sensor to be used by a Palm for automated dialing. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk fault tolerance and a embedded hardware solution.....??
Hello, IMHO you have a problem with the hardware that Asterisk runs on. You should really look around because there are a number of companies selling intel based systems with a cPCI bus fully hot swap capable. I think the only problem would be getting network adapters compatible with * but then this is only a problem of drivers easily solved by a good programmer. If you test out Asterisk on a fully redundant box and you find problems I think you'd be welcome to send in a patch to fix them so that * could be used in enterprise computing instead of sending in a two page e-mail with the problems we all know about ! Regards Kiss Karoly On Thu, 4 Mar 2004, Randall Shimizu wrote: Asterisk fault tolerance and a embedded hardware solution.?? Has anyoone tried implement Asterisk as a hardware based solution similar to Soekris firewall? Asterisk fault tolerance: I ran across this posting about Asterisk and here is some interesting thoughts to ponder http://groups.google.com/groups?hl=enlr=ie=UTF-8oe=UTF-8selm=aca5dd1d9141c07addd9d3414e934380%40free.teranews.comrnum=14 Not blow anyone's ASTERISK bubble BUT,,, Show me an Asterisk system that can: 1) Have a communication bus that can survive the removal of the CPU, and still have calls in progress that remain active until the calling parties hang up. Difficult problem to solve. One would have to have some sort of parallel network connection. Perhaps one could have a buffering or cache solution. The CPU problem could be solved by a blade server or failover. 2) I have yet to hear of any Asterisk box running a fully redundant CPU configuration. I bet this is possible.Especially with the newer hot swap cPCI bus systems and slave CPU cards. Even better if the chassis has and embedded H.110, or equivalent in LAN/memory, switching bus. Yes could be solved. 3) A redundant configuration where either CPU can talk to the communications boards (T1/E1), and LAN interfaces. And which can address all boards in the system redundantly. Sounds like a job for Infiniband or a platform that has a switched crossbar architecture like IBM P-Series or Sun. 4) A redundant configuration that has either shared system memory between the CPU's, or at least table copies between memory that hold all static and dynamic call information. 5) A redundant configuration that can swap between system CPU's in less than 20 seconds. 6) A redundant configuration that can synchronize on, and share one, two , and more network clocking signals. Plus synchronize on a independent stratum 3 or greater clock source. 7) And can support 1,000 or more endpoints (TDM and/or IP) without choking on it's own guts. 8) A redundant configuration that can synchronize on, and share one, two , and more network clocking signals. Well it's a lot to ask, but enterprise computing demands a lot. -- ___ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Supervised transfer (almost) with GS phone
Stephen R. Besch wrote: I have now tested a (previously suggested) method for doing supervised transfers using the Grandstream SIP phone. It isn't perfect, but it works and is very functional. Here are the steps: When I try this, all goes well until, after putting the original caller on hold and then getting a dialtone, I dial another extension, and then get these errors on the CLI: find_user: Call from user 'btel' rejected due to usage limit of 1 find_user: Call from user 'btel' rejected due to usage limit of 1 find_user: Call from user 'btel' rejected due to usage limit of 1 find_user: Call from user 'btel' rejected due to usage limit of 1 Then the Grandstream gives me a busy, and my orignal caller is a zombie. What am I doing wrong? Thx. 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
Re: [Asterisk-Users] Supervised transfer (almost) with GS phone
Brian Capouch wrote: Stephen R. Besch wrote: I have now tested a (previously suggested) method for doing supervised transfers using the Grandstream SIP phone. It isn't perfect, but it works and is very functional. Here are the steps: When I try this, all goes well until, after putting the original caller on hold and then getting a dialtone, I dial another extension, and then get these errors on the CLI: find_user: Call from user 'btel' rejected due to usage limit of 1 find_user: Call from user 'btel' rejected due to usage limit of 1 find_user: Call from user 'btel' rejected due to usage limit of 1 find_user: Call from user 'btel' rejected due to usage limit of 1 Then the Grandstream gives me a busy, and my orignal caller is a zombie. What am I doing wrong? Thx. 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 check sip.conf: incominglimit=1 outgoinglimit=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem to place calls to NIKOTEL
Phillipp, thank you for your help. At 00:57 04.03.2004 +0100, Philipp von Klitzing wrote: Hi! exten = _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r) Use this instead: exten = _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r) this works, in the meantime I do understand also why it did send the wrong call setup packet. The not working Dial command did not take the variables form the context nikotel in the sip.conf, because this context was not called. I was a bit confused by the examples. jakob ___ 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] how to disable zap debug!!!
how to disable this DEBUG information... I am getting this on Asterisk CLI --- Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception: Exception on 22, channel 4 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:2765 zt_handle_event: Got event On hook(1) on channel 4 (index 0) Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:1133 zt_disable_ec: disabled echo cancellation on channel 4 Mar 5 16:18:17 DEBUG[426001]: channel.c:2275 ast_channel_bridge: Didn't get a frame from channel: Zap/4-1 Mar 5 16:18:17 DEBUG[426001]: channel.c:2343 ast_channel_bridge: Bridge stops bridging channels Zap/3-1 and Zap/4-1 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:1715 zt_hangup: Hangup: channel: 4 index = 0, normal = 22, callwait = -1, thirdcall = -1 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:1133 zt_disable_ec: disabled echo cancellation on channel 4 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:2095 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/4-1 Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:1076 update_conf: Updated conferencing on 4, with 0 conference users -- Hungup 'Zap/4-1' --- -- Atif Rasheed Convergence (Business Systems) http://www.convergence.com.pk -- ___ 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] Can't find capi.conf syntax to use 2 controllers
Hello, I'm trying to configure asterisk with 2 controller (one is a AVM fxusb, the other is AVM fcpci). Each controller is bound to a separate BRI ISDN line. The two modules are correctly loaded and configured by capinit: # capiinit status 1 fxusb running fritz-usbA1 3.10-02 2 2 fcpci running fritz-pciA1 3.11-02 0xe800 10 I've configure capi.conf as follows: [interfaces] msn=041534,041534 incomingmsn=* controller=1,2 softdtmf=0 devices=2,2 Asterisk reports: *CLI capi info Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. So, apparently everthing is fine, but incoming calls form second msn are not detected by asterisk. I not able to find where support for multiple controllers is documented. May anyone help please? Thanks in advance. Luca Azzalini ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to disable zap debug!!!
On Fri, 2004-03-05 at 12:18, atif wrote: how to disable this DEBUG information... I would have intuitively said 'zap no debug' but apparently the 'no debug' is not implemented for zap although it exist for sip, iax, h323, skinny and mgcp. Should we consider this absence as a bug worthy of a wishlist item ? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] how to disable zap debug!!!
On 2004 Mar 05, at 05:18, atif wrote: how to disable this DEBUG information... I am getting this on Asterisk CLI --- Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception: Exception on 22, channel 4 Edit /etc/asterisk/logger.conf. On the line beginning with console, remove the debug item, then issue the command logger restart. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] segfault and backtrace info
On 2004 Mar 04, at 16:12, Philipp von Klitzing wrote: #0 0x080570a6 in ast_queue_frame (chan=0x810a770, fin=0x41dfd0cc, lock=1) at channel.c:368 368 cur = chan-pvt-readq; (gdb) bt Post a 'bt full' to bugs.digium.com. -Tilghman ___ 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] ?Application Hardware Recommendation
Greetings All! I just found Asterisk and am new this list. I have a home office that I have an old DOS based IVR Telephone System (Telepro) running on an old 386 pc and Dialogic D41D board. I would like to upgrade to something new and came across Asterisk. I have read the documentation and it does appear very flexible, but I thought I would run my questions by the list, before I jump in. My current setup looks something like this. Extensions When someone enters an extension it will either be routed to an outside number, for example Joe is at ext 400. Caller enters 400 and is placed on hold while phone system dials Joe's Cell phone if Joe does not answer phone system tries his home number. If Joe answers, phone system announce, I have a call for you, Press 1 to Accept Press 2 To Decline. Joe presses 1, call is transferred to him. Joe presses 2, call is sent to next number, if no further numbers, call is sent vm. Or virtually, for example Caller presses 1 for general sales. Screen Prompt appears stating Sales Call Accept/Busy. Pickup phone click accept and call is released. Click busy or prompt times out (no response) call is routed to next number or to vm Does Asterisk make sense as a replacement for this system? What hardware would you recommend? I have two incoming centrix lines. Thanks for any info. ~Hopper ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to disable zap debug!!!
Tilghman Lesher wrote: On 2004 Mar 05, at 05:18, atif wrote: how to disable this DEBUG information... I am getting this on Asterisk CLI --- Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception: Exception on 22, channel 4 Edit /etc/asterisk/logger.conf. On the line beginning with console, remove the debug item, then issue the command logger restart. -Tilghman That should be logger reload and not restart. John ___ 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] H323 termination to Cisco 5300
Hi, While terminating calls to Cisco 5300 the called party hears converstion all OK. However, calling party hears periodic short bursts of interferance and/or lost packets noise. I can see on CLI this: Mar 5 14:35:53 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference is 3760, ms is 490 Mar 5 14:35:53 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference is 3200, ms is 420 Mar 5 14:35:55 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference is 704, ms is 108 Mar 5 14:35:55 DEBUG[458773]: rtp.c:943 ast_rtp_raw_write: Difference is 952, ms is 139 And so on... I have come across above message before but I just can not remember what it is for... Any pointers would be greatly appreciated. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to disable zap debug!!!
On 2004 Mar 05, at 08:28, John Fraizer wrote: Tilghman Lesher wrote: On 2004 Mar 05, at 05:18, atif wrote: how to disable this DEBUG information... I am getting this on Asterisk CLI --- Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception: Exception on 22, channel 4 Edit /etc/asterisk/logger.conf. On the line beginning with console, remove the debug item, then issue the command logger restart. That should be logger reload and not restart. Ah, yes. Blame me for always typing logtabretabenter -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3com NBX phones
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Bougues Sent: Friday, March 05, 2004 3:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3com NBX phones [...] Note that the hardware is probably not the same as the standard NBX phones : my SIP phones did feature an IR sensor to be used by a Palm for automated dialing. That's actually an option on the better NBX phones...your is probably a 2102-IR or similar, and has been since at least when I did my last NBX rollout about a year and a half ago. What seems different is that you could flash it at all. When connecting to an NBX, these phones grab their firmware from the NBX they pin up to. I suppose there is a flashable area on the phone that is used as a boot loader in NBX mode, and probably to store the whole image when flashed with SIP. Can anyone confirm these are the same phones? Because I still have boxes of them somewhere too (that seems to be a common thread here). Daryl ___ 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] Ahead SIPPS and Asterisk
Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it to register, but when I dial into a number as soon as voice is to be connected I get a Warning 399 SDP body missing message and then a BYE disconnecting the call. The setup I have works great with Xten's x-pro, but can't get it to work with SIPPS. Any hints? ___ 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] OT: Snom 105
Greets, Apologies for the OT post. I'm working with a Snom 105 and can't seem to find the Administrator's Manual for this phone on Snom's website. Does anyone know where to find this document? Anyone know how to perform a factory reset on this device? After upgrading the firmware to 2.03o, it appears that I'm locked out of the administrative menu. -- Eric Hendrickson Sr. Solutions Analyst Iophase Inc. 19200 Von Karman, Ste. 400 Irvine, California 92612 949-608-1770 x202 888-627-6273 signature.asc Description: OpenPGP digital signature
Re: [Asterisk-Users] 3com NBX phones
The IR device is a 3rd-party piece of hardware from Extended System (now owned by iFoundry). The SIP phone looks like all of the other 3com IP phones that I have seen and turning it over with the front of the phone facing up the connectors go from left to right as follows: 1. Handset connector 2. IRDA (serial) RJ-45 connector 3. PC Ethernet RJ-45 connector 4. Wall Ethernet RJ-45 connector 5. Power adapter Maybe this will help in comparing the units. I have posted my last SIP firmware (with appropriate disclaimers) to the list but it is held up in moderator no man's land. [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Bougues Sent: Friday, March 05, 2004 3:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3com NBX phones [...] Note that the hardware is probably not the same as the standard NBX phones : my SIP phones did feature an IR sensor to be used by a Palm for automated dialing. That's actually an option on the better NBX phones...your is probably a 2102-IR or similar, and has been since at least when I did my last NBX rollout about a year and a half ago. What seems different is that you could flash it at all. When connecting to an NBX, these phones grab their firmware from the NBX they pin up to. I suppose there is a flashable area on the phone that is used as a boot loader in NBX mode, and probably to store the whole image when flashed with SIP. Can anyone confirm these are the same phones? Because I still have boxes of them somewhere too (that seems to be a common thread here). Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3com NBX phones
On Fri, Mar 05, 2004 at 10:25:49AM -0500, Clif Jones wrote: The IR device is a 3rd-party piece of hardware from Extended System (now owned by iFoundry). The SIP phone looks like all of the other 3com IP phones that I have seen and turning it over with the front of the phone facing up the connectors go from left to right as follows: 1. Handset connector 2. IRDA (serial) RJ-45 connector 3. PC Ethernet RJ-45 connector 4. Wall Ethernet RJ-45 connector 5. Power adapter Maybe this will help in comparing the units. I have posted my last SIP firmware (with appropriate disclaimers) to the list but it is held up in moderator no man's land. How about part numbers, model numbers, revisions, etc, from the labelling on the phone? Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. We aren't surrounded. We're in a target-rich environment. ___ 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] Not able to dial 9 to get out with SIP Grandstream BudgeTone-100 or SIP softphone
Hi everyone, I am having problems dialing 9 to get an external line with my SIP phones or SIP clients. I have been looking for months on websites, sitting in MIRC rooms, and reading * documentation but I cannot seem to find a solution. My asterisk box is sitting directly on the internet ( NO NAT ) with a firewall. I have also tested this box on my LAN and I have the same issue ( this is not a firewall issue ). I am using a T-100P card and an Adtran Total Access unit for all my analog phones which for now is all I use. My Grand stream SIP phone works fine for calling internal extensions with no problems at all. When I try and dial 9 and a number, after a wait of a few seconds I get 404 displayed on the screen and a busy signal. I have tried to tweak everything I know within the dial plan, but I always seem to have the same issue. I previously tried to attach my sip and extensions.conf but the email is too big for the mailing list. I have pasted small sections of them below. Id very much appreciate any help anyone can provide. SIP Conf [gs01] type=friend username=gs01 secret=pass nat=1 host=dynamic qualify=yes dtmfmode=info canreinvite=no EXTENSIONS.CONF [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp TRUNK=Zap/g2 RINGOUT=Zap/14Zap/7Zap/8Zap/9Zap/10Zap/11Zap/12 [trunkint] exten = _9011.,1,Dial(${TRUNK}/www${EXTEN:1}) exten = _9011.,2,Congestion [trunkld] exten = _91NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1}) exten = _91NXXNXX,2,Congestion [trunklocal] exten = _9NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1}) exten = _9NXXNXX,2,Congestion exten = 9411,1,Dial(${TRUNK}/www${EXTEN:1}) exten = 9411,2,Congestion exten = 9911,1,Dial(${TRUNK}/www${EXTEN:1}) exten = 9911,2,Congestion [local] ;trusted users only! ignorepat = 9 include = default include = parkedcalls include = trunklocal include = trunktollfree include = trunkint include = trunkld include = phones include = voicemail include = recording [macro-stdexten] exten = s,1,Dial(${ARG2},20) exten = s,2,Voicemail2(u${ARG1}) exten = s,3,Goto(default,s,1) exten = s,102,Voicemail2(b${ARG1}) exten = s,103,Goto(default,s,1) [phones] exten = 200,1,Macro(stdexten,200,Zap/10) ;SIP phones ;Grandstream Phones exten = 210,1,Dial(SIP/gs01) exten = 222,1,Dial(SIP/bradwell) exten = _64xx,1,Dial(SIP/gs${EXTEN:2}|20) exten = _64xx,2,Voicemail2(u${ARG1}) exten = _64xx,3,Congestion exten = _64xx,102,Voicemail2(b${ARG1}) exten = _64xx,103,Congestion [sipstart] include = phones include = voicemail include = default include = trunklocal include = trunktollfree Thanks, Steve [EMAIL PROTECTED]
Re: [Asterisk-Users] OT: Snom 105
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Thanks Sven. I think I've made things worse now :-0 Playing around with some key combinations while booting the phone, I think I've disabled the display. I see the Snom logo briefly at the bootloader but after that nothing is displayed on the panel. The backlight comes on when I pick up the receiver and the phone does work--just no display. Any suggestions? ... Other than to quit mucking around in places unknown. Sven Fischer wrote: | Hi, | | the default value to come back to admin mode for all snom phones is (for | times zero). You are right, there is an admin manual for snom200 available | only at the moment. Please see that for questions. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFASKlDbziSUSbgS2kRAllhAJ4+yp0V4EdALVHn3FN1VwIPa7HFswCeOPyz 1yF8nwRBGm+XVUvWAxS6R+E= =+yu1 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Snom 105
I could not find one either, but I did find one for the 200 series phones, which seemed similar. That is where I found the default admin password once I locked myself out. Check the 200 series product page. On Fri, 2004-03-05 at 07:18, Eric Hendrickson wrote: Greets, Apologies for the OT post. I'm working with a Snom 105 and can't seem to find the Administrator's Manual for this phone on Snom's website. Does anyone know where to find this document? Anyone know how to perform a factory reset on this device? After upgrading the firmware to 2.03o, it appears that I'm locked out of the administrative menu. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] flash button on GS101
On Thu, 2004-03-04 at 23:21, Steven Critchfield wrote: On Fri, 2004-03-05 at 02:04, dkwok wrote: Has anyone using the flash button on GS101 to access call waiting? My experience is that it does not work. I read in the list that it may need to tweak the flash duration to under 100msec. Has anyone have any solution? GS, as in Grandstream Budgetone? If so, then the flash is sent out of band. It has been mentioned that the quality can be questionable and the flash button may not be functional due to quality control issues. Please verify with a sniff of the wire using something like ethereal to verify whether or not the button press actually hits the wire in a meaningful manner. This is the end of what my GS is sending when I press flash so at least the button is working :- INFO..Use r-Agent: Grandstream BT100 1.0.4.46..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIB E..Content-Type: application/dtmf-relay..Content-Length: 25Signal=16..Duration=40875 but is it meaningful? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Snom 105
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Yes, I can get to the webpage. And the phone plays its happy tune after it boots up. Just no display on the LCD panel. Sven Fischer wrote: | Eric, | | the phone is working ? Can you access the webpage ? | | regards, | | Sven | | On Friday 05 March 2004 17:22, Eric Hendrickson wrote: | |Thanks Sven. I think I've made things worse now :-0 Playing around with |some key combinations while booting the phone, I think I've disabled the |display. I see the Snom logo briefly at the bootloader but after that |nothing is displayed on the panel. The backlight comes on when I pick up |the receiver and the phone does work--just no display. | |Any suggestions? ... Other than to quit mucking around in places unknown. | |Sven Fischer wrote: || Hi, || || the default value to come back to admin mode for all snom phones is | | (for | || times zero). You are right, there is an admin manual for snom200 | |available | || only at the moment. Please see that for questions. | |___ |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 | | - -- Eric Hendrickson Sr. Solutions Analyst Iophase Inc. 19200 Von Karman, Ste. 400 Irvine, California 92612 949-608-1770 x202 888-627-6273 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFASLQbbziSUSbgS2kRAt+WAJ4yrw6libJTqJD/czxo8w7HAe+FmwCfcO0y p7MDqoiM0rfSX0pvJAFy97k= =G0u6 -END PGP SIGNATURE- ___ 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] Kernel - TE410P
Hi everyone, Is there something that needs to be done with RedHat 9 kernel prior to installing TE410P and loading wct4xxp zap module? I mean, is there a kernel patch or something else that must be installed, or is it enough just to compile zaptel driver? thanks Tomica
[Asterisk-Users] dropped calls
Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Kernel - TE410P
compiling zaptel is enough, I have one and it loaded up just fine on RH9, make sure you modprobe zaptel, modprobe wct4xxp and then ztcfg -vvv before you try starting the first time. MATT--- -Original Message- From: Tomica Crnek [mailto:[EMAIL PROTECTED] Sent: Friday, March 05, 2004 12:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Kernel - TE410P Hi everyone, Is there something that needs to be done with RedHat 9 kernel prior to installing TE410P and loading wct4xxp zap module? I mean, is there a kernel patch or something else that must be installed, or is it enough just to compile zaptel driver? thanks Tomica ___ 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] SIP = Zaptel TDM400P issue
Situation: SIP phone A calls Asterisk. Asterisk forwards to another SIP agent B. The SIP agent B forwards A back to an Asterisk extension that is mapped to a TDM400P channel Once all of this has transpired, there is no audio channel between the SIP phone A and the TDM400P. If A is not registered with Asterisk performs the same sequence, a half- duplex connection results in which A can hear the TDM400P, but not the other way around. If the above procedure is repeated in reverse, that is: TDM400P dials the extension of SIP agent B. SIP agent B forwards TDM400P to the asterisk extension of SIP A. A full duplex connection results, and everything is normal. Any ideas? Elaborations needed? ===sip.conf=== [general] port = 5060 ; Port to bind to bindaddr = 192.168.0.104; Address to bind to context = from-sip-internal disallow=all allow=ulaw [201] username=201 type=friend secret=password host=dynamic dtmfmode=inband === 201 is SIP phone A. -Dustin Mulcahey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not able to dial 9 to get out with SIP Grandstream BudgeTone-100 or SIP softphone
From a brief look, it seems you do not have a context= in your sip.conf file for the extension. If you don't put a contxt in, I don't know what it assumes, and it will not include the contexts you have set up to define external access. From looking at your dialplan, if you put context=local into the [gs01] entry in sip.conf, you should be able to make outbound calls from this extension, as you will be forced into 'local' context and will be able se see all the external access contexts you have defined. Let us know how you get on... Rgds Tim Robinson Basingstoke, UK Stephen Foster wrote: Hi everyone, I am having problems dialing 9 to get an external line with my SIP phones or SIP clients. I have been looking for months on websites, sitting in MIRC rooms, and reading * documentation but I cannot seem to find a solution. My asterisk box is sitting directly on the internet ( NO NAT ) with a firewall. I have also tested this box on my LAN and I have the same issue ( this is not a firewall issue ). I am using a T-100P card and an Adtran Total Access unit for all my analog phones which for now is all I use. My Grand stream SIP phone works fine for calling internal extensions with no problems at all. When I try and dial 9 and a number, after a wait of a few seconds I get 404 displayed on the screen and a busy signal. I have tried to tweak everything I know within the dial plan, but I always seem to have the same issue. I previously tried to attach my sip and extensions.conf but the email is too big for the mailing list. I have pasted small sections of them below. Id very much appreciate any help anyone can provide. SIP Conf [gs01] type=friend username=gs01 secret=pass nat=1 host=dynamic qualify=yes dtmfmode=info canreinvite=no EXTENSIONS.CONF [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp TRUNK=Zap/g2 RINGOUT=Zap/14Zap/7Zap/8Zap/9Zap/10Zap/11Zap/12 [trunkint] exten = _9011.,1,Dial(${TRUNK}/www${EXTEN:1}) exten = _9011.,2,Congestion [trunkld] exten = _91NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1}) exten = _91NXXNXX,2,Congestion [trunklocal] exten = _9NXXNXX,1,Dial(${TRUNK}/www${EXTEN:1}) exten = _9NXXNXX,2,Congestion exten = 9411,1,Dial(${TRUNK}/www${EXTEN:1}) exten = 9411,2,Congestion exten = 9911,1,Dial(${TRUNK}/www${EXTEN:1}) exten = 9911,2,Congestion [local] ;trusted users only! ignorepat = 9 include = default include = parkedcalls include = trunklocal include = trunktollfree include = trunkint include = trunkld include = phones include = voicemail include = recording [macro-stdexten] exten = s,1,Dial(${ARG2},20) exten = s,2,Voicemail2(u${ARG1}) exten = s,3,Goto(default,s,1) exten = s,102,Voicemail2(b${ARG1}) exten = s,103,Goto(default,s,1) [phones] exten = 200,1,Macro(stdexten,200,Zap/10) ;SIP phones ;Grandstream Phones exten = 210,1,Dial(SIP/gs01) exten = 222,1,Dial(SIP/bradwell) exten = _64xx,1,Dial(SIP/gs${EXTEN:2}|20) exten = _64xx,2,Voicemail2(u${ARG1}) exten = _64xx,3,Congestion exten = _64xx,102,Voicemail2(b${ARG1}) exten = _64xx,103,Congestion [sipstart] include = phones include = voicemail include = default include = trunklocal include = trunktollfree Thanks, Steve[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ahead SIPPS and Asterisk
What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired? Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose use authentication, fill in the blanks, I found also putting the realm in made it work correctly. Hope that helps in some way. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlton J. O'Riley Sent: 05 March 2004 15:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ahead SIPPS and Asterisk Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it to register, but when I dial into a number as soon as voice is to be connected I get a Warning 399 SDP body missing message and then a BYE disconnecting the call. The setup I have works great with Xten's x-pro, but can't get it to work with SIPPS. Any hints? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 vs. G.729 pass thru
Derek, Can you use fax with G.729? I know that only ULAW codec can use for fax but I don't know that if you can use fax with G.729 or not. BTW, what service provider that you are using? Quality can sometime depend on provider too. Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 9:12 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru On all tests, I've run, fantastic. I haven't had any issues with voice quality at all, even on analog lines. Derek -Original Message- From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dropped calls
There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ahead SIPPS and Asterisk
I'll try to look over my config again. Not sure I put a realm in, but everything else seemed fine. I get the acquired message and I see the SIP messages flowing on the Asterisk server, but once any sound needs to be sent, it dies. I'm using g711ulaw and I was calling into an announcement menu to test that I have setup on the server. The entry in my sip.conf is exactly what I use for X-Pro, so that is why I am confused by the missing SDP message. -Original Message- Message: 8 Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk Date: Fri, 5 Mar 2004 18:25:05 - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired?=20 Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose use authentication, fill in the blanks, I found also putting the realm in made it work correctly. Hope that helps in some way. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ahead SIPPS and Asterisk
I'll try to look over my config again. Not sure I put a realm in, but everything else seemed fine. I get the acquired message and I see the SIP messages flowing on the Asterisk server, but once any sound needs to be sent, it dies. I'm using g711ulaw and I was calling into an announcement menu to test that I have setup on the server. The entry in my sip.conf is exactly what I use for X-Pro, so that is why I am confused by the missing SDP message. -Original Message- Message: 8 Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk Date: Fri, 5 Mar 2004 18:25:05 - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired?=20 Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose use authentication, fill in the blanks, I found also putting the realm in made it work correctly. Hope that helps in some way. ___ 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] Ahead SIPPS and Asterisk
Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it to register, but when I dial into a number as soon as voice is to be connected I get a Warning 399 SDP body missing message and then a BYE disconnecting the call. The setup I have works great with Xten's x-pro, but can't get it to work with SIPPS. Any hints? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dropped calls
Hello, I'll try that, but why on earth gs phones with the same firmware on another * server, work with no problem? I've failed to state I'm using zaprtc, since there is no digium hardware on the server. Does it matter? Thanks, --- Paulo. On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote: There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ 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 -- Cumprimentos, --- Paulo Loureiro Netmania ___ 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] Asterisk fault tolerance and a embedded hardware solution.....??
Hello, IMHO you have a problem with the hardware that Asterisk runs on. You should really look around because there are a number of companies selling intel based systems with a cPCI bus fully hot swap capable. I think the only problem would be getting network adapters compatible with * but then this is only a problem of drivers easily solved by a good programmer. If you test out Asterisk on a fully redundant box and you find problems I think you'd be welcome to send in a patch to fix them so that * could be used in enterprise computing instead of sending in a two page e-mail with the problems we all know about ! Ok Sorry, I if the email was a bit provocative. I was just trying to get some suggestions thoughts about hardware and fault tolerance with Asterisk. Regards Kiss Karoly On Thu, 4 Mar 2004, Randall Shimizu wrote: Asterisk fault tolerance and a embedded hardware solution.?? Has anyoone tried implement Asterisk as a hardware based solution similar to Soekris firewall? Asterisk fault tolerance: I ran across this posting about Asterisk and here is some interesting thoughts to ponder -- ___ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm ___ 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] RE: Asterisk-Users digest, Vol 1 #3012 - 11 msgs
I'm at a little bit of a loss here. I'm going to enclose my SIP output for this session that hopefully someone knows why I get the SDP not available message when using SIPPS to Asterisk. It registers great and when I call SIPPS it rings, but when it answers I get the same problem with the SDP not available message. Any help would be greatly appreciated. Thanks... - SIP Session messages --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492417237-4578560e192.168.3.69 From: sip:[EMAIL PROTECTED] ;tag=1d59b0b2 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Ahead SIPPS IP Phone Version 2.0.45.18 Expires: 180 Accept: application/ sdp Content-Type: application/ sdp Content-Length: 243 Contact: sip:[EMAIL PROTECTED] v=0 o=SIPPS 492417159 492417162 IN IP4 192.168.3.69 s=SIP call c=IN IP4 192.168.3.69 t=0 0 m=audio 1 RTP/AVP 0 8 97 2 3 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 SIP/2.0 407 Proxy Authentication Required Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492417237-4578560e192.168.3.69 From: sip:[EMAIL PROTECTED] ;tag=1d59b0b2 To: sip:[EMAIL PROTECTED] ;tag=as6b4bb6b5 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk,nonce=6c08deed Content-Length: 0 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: sip:[EMAIL PROTECTED] ;tag=1d59b0b2 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Proxy-Authorization: Digest username=101,realm=asterisk,uri=sip:192.168.3.69,nonce=6c08deed,nc= 0001,response=f058d443e9cebe0b23ca1adf0db21afb Content-Type: application/ sdp Content-Length: 243 Date: Fri, 05 Mar 2004 19:16:01 GMT Contact: sip:[EMAIL PROTECTED] Expires: 180 Accept: application/ sdp User-Agent: Ahead SIPPS IP Phone Version 2.0.45.18 v=0 o=SIPPS 492417159 492417162 IN IP4 192.168.3.69 s=SIP call c=IN IP4 192.168.3.69 t=0 0 m=audio 1 RTP/AVP 0 8 97 2 3 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 SIP/2.0 100 Trying Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: sip:[EMAIL PROTECTED] ;tag=1d59b0b2 To: sip:[EMAIL PROTECTED] ;tag=as176f35b3 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: sip:[EMAIL PROTECTED] ;tag=1d59b0b2 To: sip:[EMAIL PROTECTED] ;tag=as176f35b3 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/ sdp Content-Length: 235 v=0 o=root 32181 32181 IN IP4 192.168.3.53 s=session c=IN IP4 192.168.3.53 t=0 0 m=audio 16246 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 11206 RTP/AVP CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: sip:[EMAIL PROTECTED] ;tag=1d59b0b2 To: sip:[EMAIL PROTECTED] ;tag=as176f35b3 Call-ID: [EMAIL PROTECTED] CSeq: 2 CANCEL Warning: 399 SDP not available Date: Fri, 05 Mar 2004 19:16:01 GMT Content-Length: 0 User-Agent: Ahead SIPPS IP Phone Version 2.0.45.18 SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: sip:[EMAIL PROTECTED] ;tag=1d59b0b2 To: sip:[EMAIL PROTECTED] ;tag=as176f35b3 Call-ID: [EMAIL PROTECTED] CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 200 OK Via: SIP/ 2.0/ UDP 192.168.3.69 ;branch=z9hG4bKnp492418173-4587980e192.168.3.69 From: sip:[EMAIL PROTECTED] ;tag=1d59b0b2 To: sip:[EMAIL PROTECTED] ;tag=as176f35b3 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE Allow: ACK Allow: CANCEL Allow: OPTIONS Allow: BYE Allow: REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/ sdp Content-Length: 235 v=0 o=root 32181 32181 IN IP4 192.168.3.53 s=session c=IN IP4 192.168.3.53 t=0 0 m=audio 16246 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 11206 RTP/AVP SIP/2.0 200 OK Via: SIP/ 2.0/ UDP
Re: [Asterisk-Users] dropped calls
I have couple of GS phone and CISCO 7960. The funny thing is that two of that GS phone keep disconnecting and also CISCO 7960 phone keeps disconnecting. But the problem appear month ago! This is really strange! Bart Hello, I'll try that, but why on earth gs phones with the same firmware on another * server, work with no problem? I've failed to state I'm using zaprtc, since there is no digium hardware on the server. Does it matter? Thanks, --- Paulo. On Fri, 2004-03-05 at 18:49, Ross Donaldson wrote: There is new firmware that may help http://www.grandstream.com/BETATEST/. Grandstream acknowledges this problem. They say it is a codec issue with asterisk. I don't know if this update addresses this problem but it may be worth a try. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Loureiro Sent: Friday, March 05, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dropped calls Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message Didn't get a frame from channel: SIP/3805-df43, but I can't figure why. asterisk logs: - Mar 5 15:57:26 DEBUG[1116957488]: build_route: Contact hop: sip:192.168.60.106 Mar 5 15:57:26 VERBOSE[1217669936]: -- SIP/3805-df43 answered SIP/-08122450 Mar 5 15:57:26 VERBOSE[1217669936]: -- Attempting native bridge of SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:26 DEBUG[1116957488]: Stopping retransmission on '[EMAIL PROTECTED]' of Response\ 25663: Found Mar 5 15:57:26 DEBUG[1217669936]: Difference is 4032, ms is 524 Mar 5 15:57:26 DEBUG[1217669936]: Ooh, format changed from UNKN to ULAW Mar 5 15:57:38 DEBUG[1217669936]: Didn't get a frame from channel: SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: Bridge stops bridging channels SIP/-08122450 and SIP/3805-df43 Mar 5 15:57:38 DEBUG[1217669936]: find_user() - decrement outUse counter Mar 5 15:57:38 DEBUG[1217669936]: is not a local user Mar 5 15:57:38 VERBOSE[1217669936]: == Spawn extension (local, 3805, 1) exited non-zero on 'SIP/-0812245\0' - The scenario: 1 server (redhat 9), asterisk (stable) and a 2 x hisax ISDN BRI. One of the BRI boards is used to dial out (ppp) on one channel and a mgetty on the other channel. The other board is in ptp and used by *. The phones are Grandstream BT101 and Handytone and are all on a switched network (3 procurve switches, stacked). The configs are ok, since the same files on another server work ok (no dropped calls), but I can post them if needed. Any help will be greatly appreciated. Thanks in advance, --- Paulo Loureiro ___ 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 -- Cumprimentos, --- Paulo Loureiro Netmania ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple * status
Tim, It looks interesting.. Are you willing to release the source code? Robert Tim Sailer said: On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote: Since there's not too much out there, I decided to take about 2 hrs and pound something into shape for a simple status for my * server. I wrote a perl script that parsed the output of 'sip show peers', 'iax2 show peers', and 'show voicemail users' through the manager interface. It dumps the output to a few simple mysql tables, and the results are displayed on a web page. Now I can see some of the basic things. http://pbx.unslept.com/status.php Before anyone comments, I know it's rough and ugly looking, but this was just proof of concept for me, done over about 2 hours while trying to do my normal job, too. I'll keep poking at the CLI to see what other cool stuff I can pull out. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI and a SIP ATA
We are interested in deploying some ADSI phones, but we are currently using Sipura SPA-2000s. Everything I read on ADSI, says that it is just audio, and you don't need a special channel bank or anything like that. But what about through the SPA? I notice that zapata.conf appears to have an option to turn on ADSI support. This makes me think that it won't work with any other channel. I searched the mailing list, but the only thing I could find was from Oct. 2002, at there was no definitive answer. Anybody know if it will work? Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 vs. G.729 pass thru
I've had some small problems when trying to users features like AbsoluteTimeout with pass thru. Other than that sound quality has been good. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Unavailable ID Sent: Thursday, March 04, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple * status
On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote: Tim, It looks interesting.. Are you willing to release the source code? Sure. let me clean it up a bit... OK, a LOT... and finish the comments, and I'll have a download link for it sometime this weekend. I'll keep the downloadable stuff up-to-date with the running version. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ahead SIPPS and Asterisk
Your problem is what I experience with Messenger, when I call it. Unfortunately I never bothered trying to work out the problem. I like the SIPPS phone features, but it is ugly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlton J. O'Riley Sent: 05 March 2004 18:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk I'll try to look over my config again. Not sure I put a realm in, but everything else seemed fine. I get the acquired message and I see the SIP messages flowing on the Asterisk server, but once any sound needs to be sent, it dies. I'm using g711ulaw and I was calling into an announcement menu to test that I have setup on the server. The entry in my sip.conf is exactly what I use for X-Pro, so that is why I am confused by the missing SDP message. -Original Message- Message: 8 Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk Date: Fri, 5 Mar 2004 18:25:05 - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired?=20 Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose use authentication, fill in the blanks, I found also putting the realm in made it work correctly. Hope that helps in some way. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simple * status
That is pretty cool. I watched the light bulbs for a while now. This is a useful tool that has many possibilities. That Tim guy is on the phone more than my teenage daughter :) calvis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, March 05, 2004 11:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Simple * status Tim, It looks interesting.. Are you willing to release the source code? Robert Tim Sailer said: On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote: Since there's not too much out there, I decided to take about 2 hrs and pound something into shape for a simple status for my * server. I wrote a perl script that parsed the output of 'sip show peers', 'iax2 show peers', and 'show voicemail users' through the manager interface. It dumps the output to a few simple mysql tables, and the results are displayed on a web page. Now I can see some of the basic things. http://pbx.unslept.com/status.php Before anyone comments, I know it's rough and ugly looking, but this was just proof of concept for me, done over about 2 hours while trying to do my normal job, too. I'll keep poking at the CLI to see what other cool stuff I can pull out. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- [This E-mail scanned for viruses by Virus Hunter at itechgroup.com] --- [This E-mail scanned for viruses by Virus Hunter at itechgroup.com] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simple * status
Nice one thanks for sharing, I look forward to it. This will be very handy for SIP call transfers. At the moment I blindly transfer on sip. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: 05 March 2004 19:49 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Simple * status On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote: Tim, It looks interesting.. Are you willing to release the source code? Sure. let me clean it up a bit... OK, a LOT... and finish the comments, and I'll have a download link for it sometime this weekend. I'll keep the downloadable stuff up-to-date with the running version. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA 200 Fax
Is anyone presently using the Sipura SPA 2000 for faxing? I was about to look into it and just figured that I would ask to see if anyone ran into any snags, problems, etc. Thanks. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA 200 Fax
On Fri, 5 Mar 2004, Mark Messmore, Technical Support, University Telcom Inc. wrote: Is anyone presently using the Sipura SPA 2000 for faxing? I was about to look into it and just figured that I would ask to see if anyone ran into any snags, problems, etc. Thanks. I'm really trying. :) It works, sort of. Basically, about 1 in 4 faxes are going out without errors. Of course, that's to an IAX peer, so I'm not sure if it's a problem with the IAX peer or with the Siupra. I'm planning on testing with my Vonage line using a X100P, and see if I get better results that way. (I can hook up to the Cisco ATA186 Vonage gave me and fax without a problem.) | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA 200 Fax
hi It works, sort of. Basically, about 1 in 4 faxes are going out without errors. Of course, that's to an IAX peer, so I'm not sure if it's a problem with the IAX peer or with the Siupra. check you IAX connection. perhaps is using gsm and that could explain the failure Faxes must be sent uncompressed, ie with [u-a]law as codecs. I have 2 fax machines over SIP here (ulaw) and never missed an hit :) -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 vs. G.729 pass thru
Hi Wes, Do you need to buy license when you are using pass thru. How does it work? I'm thinking about using pass thru for voip since the service provider has g.279 codec. Can you setup your * box connects to telco termination with pass thru? PBX =[T100P]= ASTERISK (*) =[G.729]= VOIP TERMINATION Thanks. - Original Message - From: Wes Marderness [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 05, 2004 11:42 AM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru I've had some small problems when trying to users features like AbsoluteTimeout with pass thru. Other than that sound quality has been good. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Unavailable ID Sent: Thursday, March 04, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple * status
I am also looking forward to it. It looks really nice! bart - Original Message - From: Craig Waddington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 05, 2004 4:57 PM Subject: RE: [Asterisk-Users] Simple * status Nice one thanks for sharing, I look forward to it. This will be very handy for SIP call transfers. At the moment I blindly transfer on sip. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: 05 March 2004 19:49 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Simple * status On Fri, Mar 05, 2004 at 08:27:59PM +0100, [EMAIL PROTECTED] wrote: Tim, It looks interesting.. Are you willing to release the source code? Sure. let me clean it up a bit... OK, a LOT... and finish the comments, and I'll have a download link for it sometime this weekend. I'll keep the downloadable stuff up-to-date with the running version. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA 200 Fax
On Fri, 5 Mar 2004, Brancaleoni Matteo wrote: check you IAX connection. perhaps is using gsm and that could explain the failure Faxes must be sent uncompressed, ie with [u-a]law as codecs. Forgot to mention that - I've tried both ulaw and alaw all the way through (sipura - asterisk, and asterisk - IAX peer). I have 2 fax machines over SIP here (ulaw) and never missed an hit :) Just curious, which SIP peers (if they are public peers)? Is this over the 'net? | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA 200 Fax
yes be sure you are using ULAW and I found that 9600 was the baud rate to go with. 14400 seemed to be unreliable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Friday, March 05, 2004 2:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura SPA 200 Fax Is anyone presently using the Sipura SPA 2000 for faxing? I was about to look into it and just figured that I would ask to see if anyone ran into any snags, problems, etc. Thanks. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic supported well?
I'm new to asterisk and quite impressed by the feature list. I have a D/4PCI already in hand. Is there any reason NOT to use this and buy a digium card instead? I basically want to set up a couple line analog system to check it out and probably use as a a Soho setup for VM, access to a postgres database, and to play with the VOIP stuff. TIA, Alfred Werner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 vs. G.729 pass thru
Yes, I do something like that. MediatrixFXO(1204)-Asterisk-MediatrixFXO(1204), I have bought license from diguim for G.729. I do not really have a telco provider just an ISP. I use if for a private network. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Unavailable ID Sent: Friday, March 05, 2004 3:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Hi Wes, Do you need to buy license when you are using pass thru. How does it work? I'm thinking about using pass thru for voip since the service provider has g.279 codec. Can you setup your * box connects to telco termination with pass thru? PBX =[T100P]= ASTERISK (*) =[G.729]= VOIP TERMINATION Thanks. - Original Message - From: Wes Marderness [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 05, 2004 11:42 AM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru I've had some small problems when trying to users features like AbsoluteTimeout with pass thru. Other than that sound quality has been good. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Unavailable ID Sent: Thursday, March 04, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA 200 Fax
On Fri, 5 Mar 2004, Justin Carlson wrote: yes be sure you are using ULAW and I found that 9600 was the baud rate to go with. 14400 seemed to be unreliable. Are you actually faxing over the 'net? I'm using ULAW both from the Sipura - Asterisk and from Asterisk - NuFone (IAX), and have the modem locked down to 9600, but most faxes still fail. :( | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic supported well?
Alfred, I am in similar position. I took the route of going with Digium boards for two primary reasons - they have excellent quality and offer the best in customer and technical support. I am starting with an X100P just for testing everything out (using IAXClient and Asterisk). We have two Network Operation Centers - one running a T-3 and the other an OC-3. We are looking to add VoIP and PABX solutions to our product line. Once we can gain a better understanding with Asterisk and the Digium card, we are upgrading to a TE405 with PRI (4 lines). Digium is extremely knowledgeable and continues to provide us with valuable information. Good luck - call the guys at Digium (Malcolm or Greg) - they are very helpful kaydon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alfred Werner Sent: Friday, March 05, 2004 5:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dialogic supported well? I'm new to asterisk and quite impressed by the feature list. I have a D/4PCI already in hand. Is there any reason NOT to use this and buy a digium card instead? I basically want to set up a couple line analog system to check it out and probably use as a a Soho setup for VM, access to a postgres database, and to play with the VOIP stuff. TIA, Alfred Werner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P / E1 dial out
Hello *, my setup of an Asterisk box with a TDM400P and an E100P went just fine except that I cannot manage to place outgoing calls via the E1 interface (while incoming calls _do_ work). The CLI says: -8- Mar 6 00:42:25 DEBUG[-1147995216]: chan_sip.c:3593 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;line=jet7pbic -- Executing Dial(SIP/265-7cf2, Zap/1/08948999807) in new stack -- Called 1/08948999807 Mar 6 00:42:25 DEBUG[-1263383632]: rtp.c:1008 ast_rtp_write: Ooh, format changed from UNKN to ALAW -- Channel 1, span 1 got hangup Mar 6 00:42:26 WARNING[-1263383632]: app_dial.c:337 wait_for_answer: Unable to forward voice Mar 6 00:42:26 DEBUG[-1263383632]: chan_zap.c:2185 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Mar 6 00:42:26 DEBUG[-1263383632]: chan_zap.c:1715 zt_hangup: Hangup: channel: 1 index = 0, normal = 19, callwait = -1, thirdcall = -1 Mar 6 00:42:26 DEBUG[-1263383632]: chan_zap.c:1133 zt_disable_ec: disabled echo cancellation on channel 1 Mar 6 00:42:26 DEBUG[-1263383632]: chan_zap.c:2095 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 Mar 6 00:42:26 DEBUG[-1263383632]: chan_zap.c:1076 update_conf: Updated conferencing on 1, with 0 conference users Mar 6 00:42:26 DEBUG[-1263383632]: chan_zap.c:2179 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Mar 6 00:42:26 DEBUG[-1263383632]: chan_zap.c:1133 zt_disable_ec: disabled echo cancellation on channel 1 -- Hungup 'Zap/1-1' -8- corresponding configs run like: ---zaptel.conf--- span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxoks=32-35 loadzone=us defaultzone=us ---/zaptel.conf--- ---zapata.conf--- [channels] context=default language=de switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 musiconhold=default group=1 callgroup=1 pickupgroup=1 immediate=no callerid=Zentrale (256) 36094-0 channel = 1 callerid=Joerg Czaika (256) 36094-245 channel = 2 callerid=Gerda Graeber-Otte (256) 36094-265 channel = 3 callerid=Petra Weigelt (256) 36094-217 channel = 4 callerid=Elfriede Gunzelmann (256) 36094-239 channel = 5 channel = 1-15,17-31 ; TDM400B section ; context=analog signalling=fxo_ks callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no usecallerid=yes callerid=analog Fax 200 channel = 32 ---/zapata.conf--- and the corresponding extension in extensions.con: -8- exten = _0., 1,Dial(Zap/1/${EXTEN}) -8- Hm, guess I just don't see the forest for the trees, do I? Any ideas? TIA, Tobias F. Leucht ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3com NBX phones
When an NBX100 is upgraded a .tar file is uploaded and installed on the box. Inside that tar file is the firmware for the phones which is downloaded when the phone boots. If someone can provide the last SIP firmware I will replace the phone firmware in the tar file with the SIP code and see if the phone can take it. I see alot of RMA's go through our office so no loss if it kills the phone and the NBX also retains its previous versions to boot to. - Original Message - From: Derek Bruce [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 4:22 PM Subject: Re: [Asterisk-Users] 3com NBX phones The 3com phones can't be flashed... they download their firmware image from the NBX call processor when they power on... However, if a SIP image you can have the phone download the image from a linux box using the bootloader provided by Tim Hogard at http://web.abnormal.com/~thogard/nbx100.shtml - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 1:54 PM Subject: Re: [Asterisk-Users] 3com NBX phones They did make a SIP phone and are about to release new SIP phones and a new product line. The old SIP phones look identical to the NBX phones but I am not sure about the guts. Possibly the 2102 could be flashed into a 1002. Here is an ebay auction but these phones are really hard to come by. I would love to hear if a 2102 can be flashed. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3082407185category=11909 - Original Message - From: Ejay Hire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 12:34 PM Subject: RE: [Asterisk-Users] 3com NBX phones The original NBX100 phones spoke a proprietary voice-over-l2 ethernet protocol, but would upgrade to ip connectivity with a liscense key on the NBX PBX box. There was an optional software package that would let the NBX talk to an h323 gateway but it ran on nt and was rather klunky. These originally came out in 97 or 98, so sip functionality in the originals is rather unlikely. -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Fugina Sent: Thursday, March 04, 2004 1:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3com NBX phones On Thu, Mar 04, 2004 at 02:38:13PM -0500, Tim Sailer wrote: Does anyone know if the 3Com NBX 2102 series phones with with * ? There are a crapload (a very precise measurement) on eBay, but I can't figure out what protocols they talk. I believe they did make a version that spoke SIP, but they're rare. The ones on eBay are most likely 100% proprietary. Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. 666,000,000 -- The number of the megabeast. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] SIP and distinctive ring
Has anyone implemented distinctive ring for SIP devices in Asterisk? My searches revealed that there was a patch created at one time but I can't tell if it was accepted or not. Basically I have a Sipura analog adapter that I would like to have ring differently for internal calls vs external calls. Thanks guys, Matt ^ ! Matt McIntyre (KF4FGZ) ! Certified Novell Administrator ! (336) 272-9139 (Campus telephone) ! (336) 215-7199 (Mobile telephone) - Please note the change ! (336) 272-9139 (Facsimile) ! E-MAIL: [EMAIL PROTECTED] ! AIM: MixMANJaVa ! ICQ: 11956085 ^ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and distinctive ring
On Fri, 2004-03-05 at 21:04, Matt McIntyre wrote: Has anyone implemented distinctive ring for SIP devices in Asterisk? My searches revealed that there was a patch created at one time but I can't tell if it was accepted or not. Basically I have a Sipura analog adapter that I would like to have ring differently for internal calls vs external calls. Thanks guys, Matt Hi Matt, Try with: exten = 1000,1,SetVar(ALERT_INFO=Bellcore-r3) -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic supported well?
Hi Alfred- I'd like to echo Kaydon's very positive comments regarding asterisk and Digium with one or two caveats. I've had lots of experience building systems with Dialogic boards (analog and E1), and more recently a few systems built with Digium's quad E1 boards (the E400P and now the TE410P), so I've had a chance to compare them under similar application load, especially high-volume IVR and also calling card apps. If you'll be running commercial apps, I would recommend that you do a lot of testing, especially load testing, with the types of applications you'll be running. Dialogic boards, although incredibly expensive, do have lots of horsepower built in for the purposes of encoding and decoding the voice streams, decoding single and DTMF codes, voice energy and cadence detection, etc. Digium boards rely 100% on the processor that they run in to perform these functions. I was a little disappointed to find that I can (so far) reliably only handle 4 E1's in a (very) high-volume IVR app. In the past I've run Dialogic-based systems which handled much more load (but also which cost 4 times as much in hardware!) Although Digium's newer TE410P board is capable of bus mastering, I found that it made little difference in the number of channels I could run. So, I guess my point is not to be over-optimistic in deciding the number of channels that you can run. Processors are relatively cheap these days, so when in doubt, opt for more processors and spread the load! I also agree that Digium's support has been great. I do wish they would spend more time in two areas: (a) doing some documentation for the boards they sell - even a one-page setup sheet would be nice (the TE410 boards arrive with nothing, not even an explanation of what the jumpers/switches on the board do)! (b) improving the an apparent bug or shortcoming in the PRI driver code, which results in framing errors not being dealt with properly. This really only effects very high volume systems, but it needs attention. (I've discussed these with Mark) Anyway, I don't want to dissuade you from choosing Digium and asterisk, they are super accomplishments, just want you to manage your expectations a bit. Test, test, test! Good luck in your projects Scott Stingel President Emerging Voice Technology Inc. Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kaydon Stanzione Sent: Friday, March 05, 2004 10:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dialogic supported well? Alfred, I am in similar position. I took the route of going with Digium boards for two primary reasons - they have excellent quality and offer the best in customer and technical support. I am starting with an X100P just for testing everything out (using IAXClient and Asterisk). We have two Network Operation Centers - one running a T-3 and the other an OC-3. We are looking to add VoIP and PABX solutions to our product line. Once we can gain a better understanding with Asterisk and the Digium card, we are upgrading to a TE405 with PRI (4 lines). Digium is extremely knowledgeable and continues to provide us with valuable information. Good luck - call the guys at Digium (Malcolm or Greg) - they are very helpful kaydon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alfred Werner Sent: Friday, March 05, 2004 5:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dialogic supported well? I'm new to asterisk and quite impressed by the feature list. I have a D/4PCI already in hand. Is there any reason NOT to use this and buy a digium card instead? I basically want to set up a couple line analog system to check it out and probably use as a a Soho setup for VM, access to a postgres database, and to play with the VOIP stuff. TIA, Alfred Werner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Internet Phone Concept Question
Hey, all. I'm new to this Asterisk stuff and have a general concept question about making calls and whatnot over the net. I have a 4-port TDM card and a 1-port x100p card for incoming. All is configured and working fine. I have a _very_ simple configuration (start simple, add bells and whistles later). I have a cable modem hook-up and access the internet with a download speed of 128k and upload speed of 32k. My Asterisk server sits on a LAN, under a Freesco router running on an Pentium I machine (10BaseT cards because). I live in Costa Rica and would like to utilize the internet, if possible, to call family and friends in the U.S.A. Can I do that with Asterisk? Can I do that with standard analog phones through Asterisk? Can I do that without having another Asterisk machine State-side? If you have a link that would explain the concepts to me, that would be fine. Or if you could kinda prime the pump for me so I can get the ball rolling on my end - that'd be very much appreciated, too. Thanks ahead of time. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie
Andrew McRory wrote: I can offer some links that helped me... [...] If anyone has other links I'd appreciate them! Don't forget http://www.asteriskdocs.org/ ! Nick ___ 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] Call roll-over question...
I have another question for the group. I'm trying to make the following happen on my Cisco phone: I have two lines configured, 2001 and 3001. If I'm talking on 2001 and someone tries to call me on 2001 I'd like the call to roll over to 3001 and then if I don't answer, it goes to Voice mail. I was able to accomplish this using the following sequence in extensions.conf (I'm doing this from memory, so I hope I got it right). exten = 2001,1,Dial(SIP/2001,20) exten = 2001,2,Dial(SIP/3001,20) exten = 2001,3,Voicemail(u2001) Now, while this works if I'm talking on 2001, the obvious problem is that if none of the extensions are busy it will ring 2001 for 4 ring, then head over to 3001 for 4 rings until going to voice mail. So, I then tried the following: exten = 2001,1,ChanIsAvail(SIP/2001) exten = 2001,2,Dial(SIP/2001,20) exten = 2001,3,Voicemail(u2001) exten = 2001,102,Transfer(3001) exten = 2001,203,Voicemail(2001) exten = 3001,1,Dial(SIP/3001,20) exten = 3001,2,Voicemail(u2001) Which seems (to my newbie eyes) that it should work, but... it doesn't. If I pickup 2001 and call from another extension it goes straight to voice mail. Both extensions (2001,3001) work on their own, so I'm certain that they are configured correctly. Also, I have call waiting shut off on the cisco phone (so it should reject the SIP call to 2001 as busy). ...any ideas? Thanks! Swannie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel on Debian
Hermann Wecke wrote: After trying and trying to compile and make Asterisk run on a Debian box, I gave up and picked another HD with RH 9 on it. No headaches. Only 1 build was necessary to build and run *. The problem I found with debian is how they decided how to do the linux header files for everyone in /usr/include/linux either libc6-dev or linux-kernel-headers packages... The fix for me since I roll my own kernels, after a lot of buggering about and head banging on the desk, the solution was rather simple... cd /usr/src ln -s linux-2.4.25 linux then just build the cvs zaptel modules as per documents... of course a couple of `uname -a` in the Makefile would have saved me all the headaches but anyways, 6 hours later and all the wiser... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ 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] Zap to SIP transfer problem
I'm having a problem with transferring a call that comes in a Zap channel and is connected with a SIP channel (on a GS HT-286). The call is answered automatically, then the user enters an extension. Dial() is called with both T and t flags. When the bridge is made between the channels, the caller on the Zap channel can hit '#' to transfer, but the caller on the SIP channel cannot. No messages whatsoever are displayed on the console when the SIP user hits any keys. What am I missing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Internet Phone Concept Question
Hi Greg, Welcome to * world :-) Your connection is slow '128k and upload speed of 32k' so you probably need the G.729 codec ($$$ - $10/channel/call from Digium). The X100P is only for dial-out from your phones that connect to TDM card. This should use to dial local number in Costa Rica. To call to US, use * to connect to the IAX service provider such as http://connect.voicepulse.com/ or http://www.nufone.net/ It looks like this: PSTN (USA) -- IAX Provider -- Asterisk -- TDM card -- Analog phones Hope this help. Tri Tu - Original Message - From: Greg Kedrovsky [EMAIL PROTECTED] To: asterisk-user [EMAIL PROTECTED] Sent: Friday, March 05, 2004 4:59 PM Subject: [Asterisk-Users] Internet Phone Concept Question Hey, all. I'm new to this Asterisk stuff and have a general concept question about making calls and whatnot over the net. I have a 4-port TDM card and a 1-port x100p card for incoming. All is configured and working fine. I have a _very_ simple configuration (start simple, add bells and whistles later). I have a cable modem hook-up and access the internet with a download speed of 128k and upload speed of 32k. My Asterisk server sits on a LAN, under a Freesco router running on an Pentium I machine (10BaseT cards because). I live in Costa Rica and would like to utilize the internet, if possible, to call family and friends in the U.S.A. Can I do that with Asterisk? Can I do that with standard analog phones through Asterisk? Can I do that without having another Asterisk machine State-side? If you have a link that would explain the concepts to me, that would be fine. Or if you could kinda prime the pump for me so I can get the ball rolling on my end - that'd be very much appreciated, too. Thanks ahead of time. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA 200 Fax
It works on grandstream handytone 286 also, i just tested both directions and it worked perfectly first time going into a fax machine and into a windows xp machine all of this over my flaky wireless link too! Panny - Original Message - From: Brancaleoni Matteo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 05, 2004 8:33 PM Subject: Re: [Asterisk-Users] Sipura SPA 200 Fax hi It works, sort of. Basically, about 1 in 4 faxes are going out without errors. Of course, that's to an IAX peer, so I'm not sure if it's a problem with the IAX peer or with the Siupra. check you IAX connection. perhaps is using gsm and that could explain the failure Faxes must be sent uncompressed, ie with [u-a]law as codecs. I have 2 fax machines over SIP here (ulaw) and never missed an hit :) -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call roll-over question...
At 05:04 PM 3/5/2004, you wrote: I have another question for the group. I'm trying to make the following happen on my Cisco phone: I have two lines configured, 2001 and 3001. If I'm talking on 2001 and snip Try this exten = 2001,1,Dial(SIP/2001SIP/3001,20) This will ring them both at the same time for 20 seconds ...any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Internet Phone Concept Question
On Fri, Mar 05, 2004 at 06:45:32PM -0800, Unavailable ID wrote: Your connection is slow '128k and upload speed of 32k' so you probably need the G.729 codec ($$$ - $10/channel/call from Digium). Yeah, I know... it's slow. But, I am in a developing country, and I I'm a tightwad (don't wanna shell out bucks for the wee bit more bandwidth if I can get by with what I got (which beats dial-up 10 ways to next Sunday). To call to US, use * to connect to the IAX service provider such as http://connect.voicepulse.com/ or http://www.nufone.net/ Thank you. It looks like this: PSTN (USA) -- IAX Provider -- Asterisk -- TDM card -- Analog phones Hope this help. Yep. This and the previous post help a lot. Thanks to both! You gave me just what I needed to start checking things out myself. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap to SIP transfer problem
Maybe you are using inband DTMF with a compressed codec. DTMF on a call with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or INFO. On Fri, 2004-03-05 at 20:39, Barton Hodges wrote: I'm having a problem with transferring a call that comes in a Zap channel and is connected with a SIP channel (on a GS HT-286). The call is answered automatically, then the user enters an extension. Dial() is called with both T and t flags. When the bridge is made between the channels, the caller on the Zap channel can hit '#' to transfer, but the caller on the SIP channel cannot. No messages whatsoever are displayed on the console when the SIP user hits any keys. What am I missing? ___ 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 -- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap to SIP transfer problem
I'm using SIP INFO and ulaw. It seems that the same thing happens from SIP to SIP as well, regardless of what the DTMF setting is. The actual problem is that the calling user can transfer, but the called user cannot. I just tried the latest CVS snapshot and the v1.0 stable branch and they both behave the same way. [EMAIL PROTECTED] wrote: Maybe you are using inband DTMF with a compressed codec. DTMF on a call with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or INFO. On Fri, 2004-03-05 at 20:39, Barton Hodges wrote: I'm having a problem with transferring a call that comes in a Zap channel and is connected with a SIP channel (on a GS HT-286). The call is answered automatically, then the user enters an extension. Dial() is called with both T and t flags. When the bridge is made between the channels, the caller on the Zap channel can hit '#' to transfer, but the caller on the SIP channel cannot. No messages whatsoever are displayed on the console when the SIP user hits any keys. What am I missing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap to SIP transfer problem
What is your ACTUAL Dial line? On Fri, 2004-03-05 at 21:19, Barton Hodges wrote: I'm using SIP INFO and ulaw. It seems that the same thing happens from SIP to SIP as well, regardless of what the DTMF setting is. The actual problem is that the calling user can transfer, but the called user cannot. I just tried the latest CVS snapshot and the v1.0 stable branch and they both behave the same way. [EMAIL PROTECTED] wrote: Maybe you are using inband DTMF with a compressed codec. DTMF on a call with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or INFO. On Fri, 2004-03-05 at 20:39, Barton Hodges wrote: I'm having a problem with transferring a call that comes in a Zap channel and is connected with a SIP channel (on a GS HT-286). The call is answered automatically, then the user enters an extension. Dial() is called with both T and t flags. When the bridge is made between the channels, the caller on the Zap channel can hit '#' to transfer, but the caller on the SIP channel cannot. No messages whatsoever are displayed on the console when the SIP user hits any keys. What am I missing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap to SIP transfer problem
exten = s,10,Dial(${ARG1}/${DIALED},19,Ttm) which translates to Dial(SIP/210-80f2, SIP/280|19|Ttm) I believe the problem is related to the Grandstream HandyTone-286. A caller can transfer, but a callee cannot. The problem does not exist with a BT101 (1.0.4.23). I just tried all of the firmware on their BETA site, from 1.0.4.35 through 1.0.4.50 and the problem was never solved. Can anyone confirm this for me? I am SO SICK of dealing with HT-286 firmware bugs! [EMAIL PROTECTED] wrote: What is your ACTUAL Dial line? On Fri, 2004-03-05 at 21:19, Barton Hodges wrote: I'm using SIP INFO and ulaw. It seems that the same thing happens from SIP to SIP as well, regardless of what the DTMF setting is. The actual problem is that the calling user can transfer, but the called user cannot. I just tried the latest CVS snapshot and the v1.0 stable branch and they both behave the same way. [EMAIL PROTECTED] wrote: Maybe you are using inband DTMF with a compressed codec. DTMF on a call with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or INFO. On Fri, 2004-03-05 at 20:39, Barton Hodges wrote: I'm having a problem with transferring a call that comes in a Zap channel and is connected with a SIP channel (on a GS HT-286). The call is answered automatically, then the user enters an extension. Dial() is called with both T and t flags. When the bridge is made between the channels, the caller on the Zap channel can hit '#' to transfer, but the caller on the SIP channel cannot. No messages whatsoever are displayed on the console when the SIP user hits any keys. What am I missing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic supported well?
Scott Stingel wrote: If you'll be running commercial apps, I would recommend that you do a lot of testing, especially load testing, with the types of applications you'll be running. Dialogic boards, although incredibly expensive, do have lots of horsepower built in for the purposes of encoding and decoding the voice streams, decoding single and DTMF codes, voice energy and cadence detection, etc. Digium boards rely 100% on the processor that they run in to perform these functions. I was a little disappointed to find that I can (so far) reliably only handle 4 E1's in a (very) high-volume IVR app. In the past I've run Dialogic-based systems which handled much more load (but also which cost 4 times as much in hardware!) Although Digium's newer TE410P board is capable of bus mastering, I found that it made little difference in the number of channels I could run. Actually, there isn't that much processing power on the Dialogic boards. They have rather limited DSP and MCU resources. That is why they only handle codecs of trivial complexity. The reason they get better high load results for IVRs is they have huge latency, which helps enormously with the response times the applications level code needs to achieve. Such high latency kills phone calls, but nobody notices for IVR use. Most Dialogic cards are incapable of doing anything other than IVR or call switching through their mezzanine buses, as they are not full duplex. Even so, the throughput you can achieve in pure IVR applications is not that great. When you hear of people with a large bunch of T1s or E1s into a Dialogic box, it is normally some limited IVR work plus a lot of call switching. That call switched data passes across the mezzanine bus, and has no impact on the main processor. Some of the JCT cards from Dialogic can be set to a lower latency, and are full duplex. This is aimed at TTS + ASR use, rather than VoIP calls. The lowest latency is still much higher than the Digium cards, but even with this the number of channels you can handle reliably is a lot lower than when you use the traditional high latency Dialogic modes. Dialogic support was great 10 years ago, but is now almost non-existant. Their drivers are buggy, and not keeping up with the times. Their previously active forums seem to be in serious decline. Most of their long term customers find it hard to say nice things about them. They are expensive. On the other hand, they do have broad approvals across the globe. I think that is their biggest asset. Regards, Steve ___ 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] gnophone and sip phone
Hi, I installed 'gnophone' on my notePC(RHL9 linux-2.4.20-30.9) with asterisk CVS-02/05/04. I have three unsolved problems: (1)call from gnophone to sip phone is OK, but gnophone's speaker volume is very low even though setting highest volume with gmix, the speaker volume is very high. The sip hardphone side: my voice returns back to earphone of handset(echo?). (2)can not make a call from sip hardphone to gnophone *CLI says as follows; Mar 6 11:27:55 NOTICE[81926]: chan_iax.c:4098 socket_read: Rejected connect attempt from 192.168.0.11, request '[EMAIL PROTECTED]' does not exist Urgent handler Mar 6 11:27:55 WARNING[81926]: chan_iax.c:3951 socket_read: Call rejected by 192.168.0.11: No such context/extension Mar 6 11:27:55 NOTICE[81926]: chan_iax.c:1050 iax_destroy: Avoiding IAX destroy deadlock -- Called [EMAIL PROTECTED] Urgent handler -- Nobody picked up in 5000 ms -- Hungup 'IAX[192.168.0.11:5036]/7' Mar 6 11:28:00 WARNING[294929]: channel.c:517 ast_channel_free: Channel 'IAX[192.168.0.11:5036]/7' may not have been hung up properly Urgent handler Mar 6 11:28:10 WARNING[294929]: pbx.c:1834 ast_pbx_run: Timeout, but no rule 't' in context 'sip' - end of *CLI - my iax.conf is; [916] type=friend host=dynamic defaultip=192.168.0.11 port=5036 secret=916 context=default my extensions.conf is; [default] .. exten = 916,1,Dial(IAX/[EMAIL PROTECTED],5,r) . What is the meaning of '[EMAIL PROTECTED]' above *CLI? Do I miss something in 'iax.conf'? (3)When I start 'gnophone', I have to do the following sequence; 1.start mpg123 some.mp3 2.start 'asterisk' 3.stop mpg123 4.start 'gnophone' Because, asterisk graps sound device and the others can not use sound device after asterisk started. How can I release 'sound device' after asterisk started? Zen ___ 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] 7960 conference ?
Anyone been able to get the conference feature on the 7960's to work without using meetme ? I get - warning, chan_sip.c:2103 process_sdp: No compatible codecs! Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 Red Alarm
Howdy - I'm trying to get a Malaysian PRI E1 up on a TE410P, with no luck. Right now, the setup is Telco - HDSL - WorldDSL UTU801- 2 BNC E1 - balun - crossover - TE410P Right now, the CSU/DSU-ish WorldDSL box has a green light indicating E1 sync, but the TE410P shows a red alarm. I checked the card by plugging the crossover from port 1 to port 2 on the 410 (it worked fine). It I change any of the cabling (i.e. swap things around), the green light goes off. I have my suspicions about the balun (http://www.ctcu.com/catalog/datacom/balun.pdf). Would a DB15F-RJ45 converter be better the the BNC-balun-RJ45 arrangement we have now? Here's my zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 The telco line IS working; it was tested and put in a couple of days ago. Any ideas why this isn't working? Nick ___ 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