Re: [Asterisk-Users] I want to display my numbers for incoming calls when some one dials my number from any where
Hi to all again,
Thanks for your quick response
as you siad set callerid via extensions.conf by using apps available. Look at
show applications via the asterisk CLI.
can i write context for incoming ,
like
(orignal)
[ukincomming]
include = defaults
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = _55.,1,Answer
exten = _55.,2,Macro(record-enable)
exten = _55.,3,Queue(dave|t|||300)
exten = _55.,4,PlayBack(vm-goodbye)
exten = _55.,5,Hangup
[macro-record-enable]
exten = s,1,AGI(set-timestamp.agi)
exten = s,2,SetVar(CALLFILENAME=${timestamp}-${MACRO_EXTEN})
exten = s,3,Monitor(wav,${CALLFILENAME})
changed
[ukincomming]
include = defaults
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = _55.,1,Answer
exten = _55.,2,Macro(record-enable)
exten = _55.,3,SetCallerID(${CALLERIDNUM}) ;i have added the following line
exten = _55.,4,Queue(dave|t|||300)
exten = _55.,5,PlayBack(vm-goodbye)
exten = _55.,6,Hangup
will it work and will display my number dialded any one. or can u
please let me know example context for this,
Cheers,
Mazhar Hussain
On Fri, 28 Jan 2005 01:22:55 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Thu, 2005-01-27 at 23:15 -0800, Mazhar Hussain wrote:
Hi to all,
I and using asterisk with following
1. TDM400p card with four FXS modules,
So there are four analog phone lines on four zap channels,
My setup is working fine.
And version is like such
Asterisk CVS-v1-0-11/27/04-20:48:45
But when some dials form his number (suppose abc) to my number
(suppose ) I get abc number on my analog phone, but now I have
purchased more than one numbers suppose xxx , ,z
. I want to change settings in extension.con or some where so that
when some one (from any number) dials x there should be x
number on analog phone and similarly if some one dials yyy from
any number there should be y on my analog phones
On analog lines, you can't specify callerid to the PSTN. You can for
internal dialing. Check out the zap conf file in /etc/asterisk/. You can
also set callerid via extensions.conf by using apps available. Look at
show applications via the asterisk CLI.
--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Caller ID in AU
Insert a Wait(2) before Answer Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Friday, 28 January 2005 17:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Caller ID in AU Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk CVS rpms for FC1 updated
ftp://ftp.linuxsys.com/pub/releases/FC1/asterisk-CVS/ This release includes the file permission corrections. feedback requested, pls. write me directly thanks. -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID in AU
Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ouch ... error while writing audio data: : Broken pipe
Title: Message Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientryJan 28 09:35:08 WARNING[3253]: loader.c:440 load_modules: Loading module pbx_realtime.so failed!Ouch ... error while writing audio data: : Broken pipe Thanks, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk HEAD - Stable schedule?
does anyone know when current HEAD is scheduled to stabilise? Is there a plan, or is it still some time in the future? I believe I saw an announcement recently that it will start stabilizing in February, with the goal of releasing 1.1 on the six-month anniversary of the 1.0 release. When was this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does asterisk support instant messaging?
Does Asterisk support Instant Messaging? How should I configure Asterisk for working as im proxy? Thanks, Paolo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP + NAT = horrible mess
Hello, I try to connect VoIP phones to Asterisk on private network, And use Asterisk as outbound proxy via his public IP. But the localnet and externip with nat=yes, just is not working, I believe it might only rewrite SIP headers but does not touch The rtp stream. Am I right ? R. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kim Lux Sent: Friday, January 28, 2005 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP + NAT = horrible mess Comments below. On Fri, 2005-01-28 at 08:18 +0800, Leo Ann Boon wrote: Kim Lux wrote: I was expecting to have to port forward too and yet our setup doesn't require it, not on the laptop nor on the wireless router. I think as long as the SIP clients open a port on the NATing device and keep them open so the SIP provider can connect to it, all is well, even if STUN isn't used. I was surprised by how easy it was to NAT the Grandstreams. I had visions of having every device being assigned a static IP and having a fistful of port forwards assigned to them on the router. You're connecting to a SIP provider or just Asterisk? Just a provider right now. I'll tackle asterisk in a few days. Most SIP provider use a far-end NAT traversal device like Jasomi, Acmepacket or Kagoo. The NAT traversal device has the intelligence to figure out the UDP port mapping used by the NAT. SER + nathelper has the effect. I guess ignorance is bliss in this case. For my SER setup, most of the time we can just plug the SIP phone into a router and it will work without any special config. Unfortunately, there're certain firewalls like PIX and MS ISA that will fail. In those cases, your best bet is to do port forwarding or use an outbound proxy. IIRC, Vonage also has the same problem. Thanks for sharing this. It may help some poor soul trying to get his SIP device working in these situations. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I want to display my numbers for incoming calls when some one dials my number from any where
Hi to all again,
Thanks for your quick response
as you siad set callerid via extensions.conf by using apps available. Look at
show applications via the asterisk CLI.
can i write context for incoming ,
like. can some one of you will guide me.
(orignal)
[ukincomming]
include = defaults
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = _55.,1,Answer
exten = _55.,2,Macro(record-enable)
exten = _55.,3,Queue(dave|t|||300)
exten = _55.,4,PlayBack(vm-goodbye)
exten = _55.,5,Hangup
[macro-record-enable]
exten = s,1,AGI(set-timestamp.agi)
exten = s,2,SetVar(CALLFILENAME=${timestamp}-${MACRO_EXTEN})
exten = s,3,Monitor(wav,${CALLFILENAME})
changed
[ukincomming]
include = defaults
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = _55.,1,Answer
exten = _55.,2,Macro(record-enable)
exten = _55.,3,SetCallerID(${CALLERIDNUM}) ;i have added the following line
exten = _55.,4,Queue(dave|t|||300)
exten = _55.,5,PlayBack(vm-goodbye)
exten = _55.,6,Hangup
will it work and will display my number dialded any one. or can u
please let me know example context for this,
Cheers,
Mazhar Hussain
- Hide quoted text -
On Fri, 28 Jan 2005 01:22:55 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Thu, 2005-01-27 at 23:15 -0800, Mazhar Hussain wrote:
Hi to all,
I and using asterisk with following
1. TDM400p card with four FXS modules,
So there are four analog phone lines on four zap channels,
My setup is working fine.
And version is like such
Asterisk CVS-v1-0-11/27/04-20:48:45
But when some dials form his number (suppose abc) to my number
(suppose ) I get abc number on my analog phone, but now I have
purchased more than one numbers suppose xxx , ,z
. I want to change settings in extension.con or some where so that
when some one (from any number) dials x there should be x
number on analog phone and similarly if some one dials yyy from
any number there should be y on my analog phones
On analog lines, you can't specify callerid to the PSTN. You can for
internal dialing. Check out the zap conf file in /etc/asterisk/. You can
also set callerid via extensions.conf by using apps available. Look at
show applications via the asterisk CLI.
--
Steven Critchfield [EMAIL PROTECTED]
___
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Asterisk-Users@lists.digium.com
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Re: [Asterisk-Users] Attended call transfer
Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Any news? I am also looking for #-Transfers for asterisk-stable. Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Continuously ringing Zap/4-1 TDM11B All of a sudden ?[Urgent Pls]
Dear All, All these days I was ahpppily using Asterisk with TDM11B, but from today all of a sudden asterisk has started acting strange. The telephone device connected to channel 1 rings continously, following info is displayed on console -- Starting simple switch on 'Zap/4-1' Jan 28 15:38:58 ERROR[77024176]: callerid.c:261 callerid_feed: fsk_serie made mylen 0 (-8) Jan 28 15:38:58 WARNING[77024176]: chan_zap.c:5414 ss_thread: CallerID feed failed: Success Jan 28 15:38:58 WARNING[77024176]: chan_zap.c:5456 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing Answer(Zap/4-1, ) in new stack Please tell me what must have gone wrong so unnoticed ? Its very urgent to put it back to work, as i dont have backup plans. I can send my config files if u wish to look at it. Eagerly awiating. Thanks in advance. Anand ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soft phone sound quality help
I have a client that experienced quality problems and he said the resolution turned out to be the QoS option for the nic card (even though their backbone didn't support QoS). Try the softphones with and without QoS to hear the difference. Anyone got any tips on improving sound quality on soft phones running under Window XP SP2? I have tried Xlite, SJPhone and Firefly. They all seem to have significant sound quality problems. We have a reasonable sized network of several hundred devices connected together using Layer 2 switches, i.e. pretty dumb switches with no QoS. I also have a Grandstream connected to the same switching gear. The Grandstream sounds pretty good with very few drop outs or sound problems on ulaw. The soft phones all have problems although they get less when going to a lower bandwidth codec, but then lower bandwidth gives you worse sound quality too. Is there any way I can improve sound quality on the softphones? Or it is pretty well the general rule that they have poor sound quality? It makes sense to install a softphone on each of the 200 desktops we have but not to buy 200 Grandstreams or equivalent, and not to upgrade all our network switches. On the Asterisk side, jitter buffer is turned on with default settings. TOS is turned on for SIP although I doubt the switches can do anything with it. I have played around with a lot of Asterisk settings but without getting good results. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CISCO 7905 Phone Weirdness
It seems on my phone, which is hooked up to a large pbx network powered by an asterisk server, that it will randomly start ringing with a callerid# of 2013 which is its username for that phone. I have looked and been watching on the asterisk command line with the -vvr switch and nothing has been seen that indicates a reason for this random ringing. This leads me to think that this trouble is not involving the asterisk server, however I am bothered by it and would like to find out what is causing the trouble. I was wondering, does anyone here have any ideas as to what might be causing the weird riging, or if not what is causing it, are there any suggestions as to what to look into to find a possible source of the problem. I don't use the 7905's, but might try: - use ethereal to see if the call is actually initiated from something external to the phone - if you are using sip firmware, implement telnet within the phone, telnet to it, and look at some of the debug options ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
Frank Sautter schrieb: * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten = _.,102,Busy() with no effect. this is the part of the extensions.conf i'm using: peter svensson gave me the hint to set priindication=outofband now i'm able to signal busy to the calling party and with setting PRI_CAUSE there are even more possibilities see http://www.voip-info.org/wiki-Asterisk+cmd+Hangup regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Command to light MWI on 7940 /7960
We have several agents on queues, and want to indicate to them that they are logged in or logged out. We have tried several different ways, from changing the screen to presenting different service menus, but cannot get anything to be in their face (their words, not mine). One of our team has suggested, as the agents do not have voicemail, is to use the MWI on the 7940 phones to indicate that they are logged in. Short of dropping a file into the INBOX of the particular extension (mucking around with files is yucky) is there any way of achieving this with the dialplan ? I was thinking of being able to send a SIP notify message oi, you at 560, light your lamp or something like that. Anyone done anything similar ? Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipua SPA-2000 and liong delay after dialling number
When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi Asterisk Server Transfers
Hi! Call is then connected as follows. PSTN - Provider - Head Office - Provider - Remote But after it is transferred, I want the resulting route to be: PSTN - Provider - Remote Otherwise Head office has 2 times the bandwidth running through it for a call not even going to one of it's own extensions. I had throught that the IAX connection between Provider and Head Office would pass off calls that way. It should - you need to find out if either the office or the provider * has notransfer=yes in the iax.conf file. If yes: remove that. Take a careful look at the IAX log and debug messages to find out when/if Asterisk tries to do a native transfer (and probably fails for some reason). Next to this you need to check that the codecs are matching, your best bet is to use the same codec on all sides. In the worst case you have exactly one of the clients talking g729 and only one * box that has a transcoding license for this. By the way, with all this I/we assume that you are using the Asterisk # transfer mechanism and not a phone-based one. Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after dialling number
The delay is a time out. The SPA does not know how many numbers it is expecting before it has a complete number for your system. The invite message is sent as a single message to asterisk containing the whole number string, as apposed to each number individually. In simple terms you have 2 options at your disposal : a) encorage users to adopt pressing gate / pound / hash (the noughts and crosses board above 9 on the keypad - i cant belive this keyboard doesn't have the symbol ;) at the end of the last digit - this in the sipura (like 99% of telephony devices) is treated as a send / termination / enter instruction and sends the instruction (invite message) to asterisk immediatly Note this only applies if your using a touch-tone / dtmf (dual-tone multi-frequency) enabled hand set. b) edit the dial plan of the sipura, to instruct the device of your dial plan, so that it understands how your system is configured. It is sensitve enough to understand that numbers like 999 / 112 / 911 are only 3 digits when national dialing is a greater length. For assitance with that google for spa-2000 user guide, which contains examples or contact me with further information of your set up Hope this helps. david On 28 Jan 2005, at 11:14, Remco Barende wrote: When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Command to light MWI on 7940 /7960
Yes. This will work provided you don't need to use the light for anything else. You are correct about sending a NOTIFY. There is a specific field you need in the message body. I think it is called mwi waiting but you should check the rfcs on this one. Sending a value yes or no will turn the light on/off respectively. Asterisk wrote: We have several agents on queues, and want to indicate to them that they are logged in or logged out. We have tried several different ways, from changing the screen to presenting different service menus, but cannot get anything to be in their face (their words, not mine). One of our team has suggested, as the agents do not have voicemail, is to use the MWI on the 7940 phones to indicate that they are logged in. Short of dropping a file into the INBOX of the particular extension (mucking around with files is yucky) is there any way of achieving this with the dialplan ? I was thinking of being able to send a SIP notify message oi, you at 560, light your lamp or something like that. Anyone done anything similar ? Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soft phone sound quality help
I've tried setting the QoS settings on the card and using the Microsoft QoS packet scheduler, in all combinations, but no changes. I don't think these applications use QoS anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, January 28, 2005 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Soft phone sound quality help I have a client that experienced quality problems and he said the resolution turned out to be the QoS option for the nic card (even though their backbone didn't support QoS). Try the softphones with and without QoS to hear the difference. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended call transfer
Thomas Dingermann wrote: Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Any news? I am also looking for #-Transfers for asterisk-stable. Thomas 1.0.x is for bug fixes only. No new features are added to 1.0.x. Blind XFER using # has been in Asterisk for a long time. It's Attended # transfers that are in CVS-HEAD only. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been quietly busy with a BRIstuffed patch that works against Asterisk Head? I also notice that I can't seem to re-compile the H323 stuff any more... with this release... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after dialling number
On Fri, 28 Jan 2005, David John Walsh wrote: The delay is a time out. The SPA does not know how many numbers it is expecting before it has a complete number for your system. The invite message is sent as a single message to asterisk containing the whole number string, as apposed to each number individually. In simple terms you have 2 options at your disposal : a) encorage users to adopt pressing gate / pound / hash (the noughts and crosses board above 9 on the keypad - i cant belive this keyboard doesn't have the symbol ;) at the end of the last digit - this in the sipura (like 99% of telephony devices) is treated as a send / termination / enter instruction and sends the instruction (invite message) to asterisk immediatly Note this only applies if your using a touch-tone / dtmf (dual-tone multi-frequency) enabled hand set. Great, thanks! This is the easiest solution, the intercom can dial a * and # I only have to terminate the number with an # :) Thanks for the tip! All my visitors at the door will be greatful :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
Hi Mark, please take a look at bristuff 0.2.0-RC5 which uses * 1.0.5: http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC5.tar.gz best regards Klaus Am Freitag, den 28.01.2005, 14:35 +0200 schrieb Mark Elkins: I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been quietly busy with a BRIstuffed patch that works against Asterisk Head? I also notice that I can't seem to re-compile the H323 stuff any more... with this release... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Command to light MWI on 7940 /7960
Thanks for that - I have got the mechanism working using system calls to
touch and remove txt files in the appropriate voicemail directories.
Is there any dialplan command to do this more elegantly ?
exten = ,1,SipNotify(${CALLERIDNUM},mwi=yes)
exten = 1112,1,SipNotify(${CALLERIDNUM},mwi=no)
I've googled and found nothing so I'm just hoping :)
Julian
Steve Blair wrote:
Yes. This will work provided you don't need to use the light for
anything else. You are correct about sending a NOTIFY. There is
a specific field you need in the message body. I think it is called
mwi waiting but you should check the rfcs on this one. Sending a
value yes or no will turn the light on/off respectively.
Asterisk wrote:
We have several agents on queues, and want to indicate to them that
they are logged in or logged out. We have tried several different
ways, from changing the screen to presenting different service menus,
but cannot get anything to be in their face (their words, not mine).
One of our team has suggested, as the agents do not have voicemail,
is to use the MWI on the 7940 phones to indicate that they are logged
in.
Short of dropping a file into the INBOX of the particular extension
(mucking around with files is yucky) is there any way of achieving
this with the dialplan ? I was thinking of being able to send a SIP
notify message oi, you at 560, light your lamp or something like that.
Anyone done anything similar ?
Julian
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[Asterisk-Users] Re: phone rings when I'm using it over VOIP - WHY?
Joseph [EMAIL PROTECTED] wrote: [snip] Do you get a call-waiting beep when you're on the phone with the original party? I think this is it, I can hear the beep so that would explain why my phone rings when I'm using it. [snip] What am I doing wrong? In your SPA-3000 setup screens, go to the Line1 page, and under Supplementary Service Settings, set Call Waiting Serv to No. That should stop it accepting a second call when one is in progress. You may need to look on the User1 page too, and set CW setting to No. These SPA-3000 units have a zillion parameters, so it's easy to miss one! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: iax.cc/sixTel local DID question
On Thu, 27 Jan 2005 22:10:43 -0500, David Mallwitz wrote: Andrew Thompson wrote: David Mallwitz wrote: Their isn't any indication of whether or not clicking the Add button will immediately add a number to my account or take me to another screen to pick a NXX. snip The form lets you choose the NXX. Actually, it didn't. I asked in a ticket what happens and the response came back that I would have gotten an email about it. I've sent the request back, so we'll see what happens. Odd. I signed up for a DID with them yesterday, and the form gave me a choice of several NXX's in my area. I think it depends upon what they have to offer in your area. In my region (Houston) they allow only the selection of one area code and nothing further. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] does asterisk support instant messaging?
Hi i was wondering the same, but one question what do you use for instant massenging, Xten Eyebean, it is so do you figure out if the video works? because the eyebean besides audio and video support instant messenging Thank You On Fri, 28 Jan 2005 10:15:43 +0100, Paolo Elefante [EMAIL PROTECTED] wrote: Does Asterisk support Instant Messaging? How should I configure Asterisk for working as im proxy? Thanks, Paolo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after dialling number
You can also adjust the Interdigit Long Timer and Interdigit Short Timer values found in the Regional settings config screen. - Pedro On Fri, 28 Jan 2005 13:36:14 +0100 (CET), Remco Barende [EMAIL PROTECTED] wrote: On Fri, 28 Jan 2005, David John Walsh wrote: The delay is a time out. The SPA does not know how many numbers it is expecting before it has a complete number for your system. The invite message is sent as a single message to asterisk containing the whole number string, as apposed to each number individually. In simple terms you have 2 options at your disposal : a) encorage users to adopt pressing gate / pound / hash (the noughts and crosses board above 9 on the keypad - i cant belive this keyboard doesn't have the symbol ;) at the end of the last digit - this in the sipura (like 99% of telephony devices) is treated as a send / termination / enter instruction and sends the instruction (invite message) to asterisk immediatly Note this only applies if your using a touch-tone / dtmf (dual-tone multi-frequency) enabled hand set. Great, thanks! This is the easiest solution, the intercom can dial a * and # I only have to terminate the number with an # :) Thanks for the tip! All my visitors at the door will be greatful :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bristuff and Realtime
Hi, I would like to use Realtime extentions with a four bri card, the classic quodbri. Normally with that card I would use * bristuffed from Klaus-Peter Junghanns, but since that package is based on stable version there is no Realtime at all in it (I suppose). Any idea, other than wait for realtime to begin stable ? :) Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Command to light MWI on 7940 /7960
There is a way to execute an external script from the exten statement
but I
don't have the command handy. My asterisk server is down right now. Check
the documentation on this command.
Asterisk wrote:
Thanks for that - I have got the mechanism working using system calls
to touch and remove txt files in the appropriate voicemail directories.
Is there any dialplan command to do this more elegantly ?
exten = ,1,SipNotify(${CALLERIDNUM},mwi=yes)
exten = 1112,1,SipNotify(${CALLERIDNUM},mwi=no)
I've googled and found nothing so I'm just hoping :)
Julian
Steve Blair wrote:
Yes. This will work provided you don't need to use the light for
anything else. You are correct about sending a NOTIFY. There is
a specific field you need in the message body. I think it is called
mwi waiting but you should check the rfcs on this one. Sending a
value yes or no will turn the light on/off respectively.
Asterisk wrote:
We have several agents on queues, and want to indicate to them that
they are logged in or logged out. We have tried several different
ways, from changing the screen to presenting different service
menus, but cannot get anything to be in their face (their words,
not mine).
One of our team has suggested, as the agents do not have voicemail,
is to use the MWI on the 7940 phones to indicate that they are
logged in.
Short of dropping a file into the INBOX of the particular extension
(mucking around with files is yucky) is there any way of achieving
this with the dialplan ? I was thinking of being able to send a SIP
notify message oi, you at 560, light your lamp or something like
that.
Anyone done anything similar ?
Julian
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--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Ouch ... error while writing audio data: : Broken pipe
Hello Stojan , This issue is related to a hardware problem (Zaptel ) . Probably , you have loaded locked zaptel modules or not loaded correctly. Try to run ztcfg and see if you have any errors on your config. Check if u have multiples instances of mpg123 ( this kind of problem .. usually leaves on the system at least 2 active processes.) I had this problem days ago ... By the way , what boards do u have there ? are u using asterisk 1.0.2 ? Regards, -Jefferson Carvalho Stojan Sljivic - Pamet wrote: Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28 09:35:08 WARNING[3253]: loader.c:440 load_modules: Loading module pbx_realtime.so failed! Ouch ... error while writing audio data: : Broken pipe Thanks, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after dialling number
Simply dial a # after the number. Remco Barende ha scritto: When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ouch ... error while writing audio data: : Broken pipe
Who has an answer for this desperate problem? file has an answer for this desperate problem! Who me? Yes you! So true!You wouldn't have happened to have downgraded from CVS head to CVS stable by any chance? Stable has no idea what realtime is... so if the old realtime modules are present, they choke and your asterisk goes kaboom. To fix this all you have to do is type rm -rf /usr/lib/asterisk/modules and then make install again in your asterisk directory. This will ensure that you have stable modules only, not a mix of head and stable.This concludes our lesson today. Please tune in tomorrow when I will make chicken 'alatwisted in the front yard.- Joshua Colp.Stojan Sljivic - Pamet wrote: Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28 09:35:08 WARNING[3253]: loader.c:440 load_modules: Loading module pbx_realtime.so failed! Ouch ... error while writing audio data: : Broken pipe Thanks, Stojan Sljivic Message sent using UebiMiau 2.7.2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipua SPA-2000 and liong delay afterdialling number
[EMAIL PROTECTED] wrote: The invite message is sent as a single message to asterisk containing the whole number string, as apposed to each number individually. Does SIP support non en-bloc dialling mode? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after diallingnumber
Pedro, You can also instruct your users to press the # key after dialing the number to get the dial to start immediately. -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] redirect different phone number to different IP phone
Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 Thanks Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Am I missing something really basichere?????helpwith Asterisk@home {Scanned} {Scanned} {Scanned}
Remember I'm new here too.
You might need to edit /etc/zaptel.conf
Check fxsks=1-4 I have four X100P cards.
If you have one X100P change it to fxsks=1
I have no idea what AMP configurator is?
David
On Thu, 2005-01-27 at 12:17 -0500, Jeff R Glassman wrote:
I also edited the Zapata.conf file I did not change the zaptel.conf, what
did you change in it
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David Shaw
Sent: Thursday, January 27, 2005 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Am I missing something really
basichere?helpwith [EMAIL PROTECTED] {Scanned} {Scanned}
I'm running [EMAIL PROTECTED] I had to edit /etc/zaptel.conf
and /etc/asterisk/zapata.conf. After that it works great.
David..
On Thu, 2005-01-27 at 09:54 -0500, dean collins wrote:
Ok, I thought the point of [EMAIL PROTECTED] was that it automatically
detected the X100P board and configured it correctly.
Is this incorrect? You still need to modify /etc/zaptel files? And not
just using the AMP configurator.
There is no mention of this on the [EMAIL PROTECTED] webpage.
Can anyone who has actually used [EMAIL PROTECTED] confirm this one way
or the other?
Thanks,
Dean
__
From:[EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of David Shaw
Sent: Thursday, January 27, 2005 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Am I missing something really basic
here?helpwith [EMAIL PROTECTED] {Scanned}
Yes, You need to add channels to your zapata.conf file.
zapata.conf
[channels]
;
; X100P plugged into PSTN
; X100P # 1
;[line1]
context=line1
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1.5
txgain=1.5
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usecallerid=yes
callerid=asreceived
channel = 1
You might need to edit /etc/zaptel.conf
Check fxsks=1-4 I have four X100P cards.
If you have one change it to fxsks=1
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface
for demo
IAXINFO=guest ; IAXtel
username/password
TRUNKL1=Zap/1
TRUNKL2=Zap/2
TRUNKL3=Zap/3
TRUNKL4=Zap/4 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[line1]
exten = s,1,Dial(SIP/101,20)
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Voicemail,101
exten = s,5,Hangup
Here I have TRUNKL1=Zap/? for each X100P cards.
[line1] tells asterisk how to answer that line.
Remember I'm very new at this, but I didn't see anyone respond to your
post.
Goog luck, David
- Original Message -
From: dean collins
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 26, 2005 5:36 AM
Subject: [Asterisk-Users] Am I missing something really basic
here? helpwith [EMAIL PROTECTED] {Scanned}
Im trying to install [EMAIL PROTECTED], Ive just downloaded
the
latest cd from soundforge. I can get it to install ok (network
card didnt auto configure but I worked out how to use
netconfig).
I worked out how to add a few grandstream budgetone fine.
Worked out how to upload music etc. Worked out how to modify
FOP.
Voicemail and meetmes work fine.
HOWEVER.
Im using a X100p. I cant get it to make a call out or use the
default extension for an incoming line.
What do I need to make the pstn connection work? Do I need to
modify Zapata.conf? there are zero instructions on the
[EMAIL PROTECTED] page as to what to do.
Can anyone help me out here.
TIA,
Dean
--
This message has been scanned for viruses and
Re: [Asterisk-Users] redirect different phone number to different IP phone
I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 Just put both incoming lines in a different context and have an extension s,1 that dials the phone you want. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STUN
I have a SER server and an * server, both have private addresses and have static nat's on the router to the internet. I have installed STUN (by vovida) on the SER server by giving the SER server a second private address on a sub interface (which is probably not right). I understand I need a public address on the SER box, however is this the correct approach to getting it working for clients behind a router e.g broadband users ? Thanks ___ ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] e164.org update
Long time coming, but we finally have a 3rd party interface on the website to add block of enum numbers in regex form... eg +4412345[678] which will match +44123456 +44123457 +44123458 also +4412345[16-18] which will match +441234516 +441234517 +441234518 or just short prefixes +4412345 so anything starting with +4412345 will match... Currently this is accessible via web interface only, but if anyone knows of any existing CSV, XML, SOAP etc interfaces that allow a similar update mechanism to other services you already use to update large numbers of routes, let us know and we will be able to quickly get something in place now that the base code exists. -- 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! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than myself. ___ E164-discuss mailing list [EMAIL PROTECTED] http://shoveler.ipl31.net/mailman/listinfo/e164-discuss ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk
VAD, Voice Activity Detection On Thu, 2005-01-27 at 21:16 -0500, Brian Dingman wrote: PCMU is g711 ULAW and PCMA is G711 ALAW. ALAW is more common in Europe. Not sure about VAD. On Thu, 27 Jan 2005 18:44:09 -0700, Kim Lux [EMAIL PROTECTED] wrote: Thanks for the tips. The Grandstream doesn't have a G711 or uLaw option for codecs. It has PCMU, PCMA and iLBC. Are any of these related to G711 ? Grandstreams have echo cancellation and it appears to be working after a few seconds of conversation. What is VAD ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ouch ... error while writing audio data: : Brokenpipe
Title: Message Hi all, Thanks for the information. Yes, I have been downgrading from HEAD to 1.0.5. I have removed the /usr/lib/asterisk/modules and I do not get previous error, but apparently a new one appeared: [cdr_tds.so]Jan 28 15:16:28 WARNING[25289]: loader.c:258 ast_load_resource: libtds.so.3: cannot open shared object file: No such file or directoryJan 28 15:16:28 WARNING[25289]: loader.c:440 load_modules: Loading module cdr_tds.so failed!Ouch ... error while writing audio data: : Broken pipe Do you know what is this related to? Regards,Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] redirect different phone number to different IP phone
Video Dery / Internet du Royaume wrote: Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers What kind of line? There has been some questions in the last day or so about DNIS, so I'm not sure that it can be done on inbound analog lines. I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 exten=5551234,1,Dial(SIP/phone1) exten=5551235,1,Dial(SIP/phone2) Customize accordingly... -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoiding queue retries without hangs?
On Thu, 2005-01-27 at 16:14 -0600, Eric Wieling wrote: You might consider upgrading to 1.0.5 release Thanks, I checked it out. With same config as for 1.0 I get: Asterisk Ready. -- Accepting AUTHENTICATED call from 192.168.0.10, requested format = 1024, actual format = 1024 -- Executing Goto(IAX2/[EMAIL PROTECTED]/2, gh-queuein|s|1) in new stack -- Goto (gh-queuein,s,1) -- Executing Queue(IAX2/[EMAIL PROTECTED]/2, ghq20) in new stack Floating point exception Not exactly an improvement. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stumped by BroadVoice SIP {Scanned}
Manjit, Do you have 3 lines with BroadVoice? If so how do you tell which
number is ring in on or which line to dial out on I have on line
with him now and would like to add two lines..
Thanks, David.
On Thu, 2005-01-27 at 14:14 -0800, Manjit Riat wrote:
I had a lot of problem with them to set up..
You need to register to sip.broadvoice.com
And need to have all of their four servers to listen to incoming calls as
ony one can send it in..
Just posted my config two days ago.
http://lists.digium.com/pipermail/asterisk-users/2005-January/085736.html
hope that helps
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 2:02 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Stumped by BroadVoice SIP
Hello guys.
I am a fairly new user to Asterisk, and I'm just having a tough time.
My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
number, and I have three systems with SIP phones.
The PBX and the SIP phones are all behind a Cisco PIX running NAT.
I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
little luck.
The SIP phones are two X-Lites on Windows and one Kphone on Linux (running
from the same system that Asterisk runs on).
It appears that the BroadVoice SIP registers and the SIP phones register,
as I can call from one Xlite to the Kphone. However, I cannot get
incoming calls from BroadVoice. Calling the BroadVoice number results in
a 'The party you wish to reach is busy and cannot...' message. I sniffed
packets and I can see packets coming in from BroadVoice on port 5060 to
the PBX, but they do not correspond with my call attempts. And debugging
the sip session shows alot of '404 Not Found'.
Also, even though this is meant as a incoming only PBX, I tried to test
outgoing calls from an X-Lite softphone to BroadVoice, but it doesn't
work, either.
I've probably screwed my configs to hell trying to get this to work, but
here they are. Any suggestions would be appreciated.
Here are my configs, decrufted...
sip.conf
[general]
context=sip
recordhistory=yes
port = 5060
bindaddr = 0.0.0.0
allow=gsm
allow=alaw
allow=ulaw
allow=adpcm
allow=speex
allow=ilbc
allow=slinear
[general]
nat=yes
register = 212999:password:[EMAIL PROTECTED]:5060
register = 212999:password:[EMAIL PROTECTED]:5060
externip = 208.59.47.2
localnet=192.168.1.0/255.255.0.0
[sip_proxy]
type=user
context=from-broadvoice
[xlite1]
type=friend
regexten=101
username=xlite1
secret=password
callerid=Stephen's Laptop 101
host=dynamic
nat=no
canreinite=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=inband
qualify=yes
[xlite2]
type=friend
regexten=103
context=sip
username=103
secret=password
callerid=Ben's Laptop 103
host=dynamic
nat=no
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=inband
quality=yes
[kphone1]
type=friend
username=kphone1
secret=password
callerid=Diablo 102
host=dynamic
allow=gsm
qualify=yes
[sip.broadvoice.com]
type=peer
host=proxy.dca.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=212999
secret=password
context=incoming
canreinvite=no
[broadvoice-out]
type=peer
dtmfmode=inband
host=147.135.0.128
user=212999
username=212999
authuser=212999
fromuser=212999
fromdomain=sip.broadvoice.com
md5secret=password
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
[broadvoice-out2]
type=peer
dtmfmode=inband
host=147.135.8.128
user=212999
username=212999
authuser=212999
fromuser=212999
fromdomain=sip.broadvoice.com
md5secret=password
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
[broadvoice-incoming]
type=peer
dtmfmode=inband
host=147.135.8.128
context=incoming
qualify=yes
nat=yes
canreinvite=no
fromdomain=sip.broadvoice.com
username=212999
fromuser=212999
insecure=very
[broadvoice-incoming2]
type=peer
dtmfmode=inband
host=147.135.0.128
context=incoming
qualify=yes
nat=yes
canreinvite=no
fromdomain=sip.broadvoice.com
username=212999
fromuser=212999
insecure=very
-
extensions.conf
-
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1; MSD digits to strip
(usually 1 or 0)
[iaxtel700]
exten = _91700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL
PROTECTED])
[iaxprovider]
[trunkint]
exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9011.,2,Congestion
[trunkld]
exten =
[Asterisk-Users] Where can I find good doc on AGI?
Hi, I have searched the list/Wiki, web and I am not able to find a decent documentation of the AGI/FastAGI interface with examples. Am I looking in wrong places? Help will be greatly appreciated. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with chan_sccp and cisco 7960
Hi !
On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone
button (or select line with line button - which automatically put second
line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable
dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5,
chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of
7960 is P00305000301.sbn.
Did anyone noticed similar problem or perhaps knows solution for this ?
Here is what asterisk puts on console (test is with 7960+7914 addon module
on 1.0.3 version of Asterisk, same thing happened on 7960 without 7914):
-
-moment of registration of telephone to asterisk:
== Got message AlarmMessage
Jan 18 16:47:09 NOTICE[23046]: sccp_actions.c:23 sccp_handle_alarm: Alarm
Message: Severity: 2, 25: Name=SEP00127FAE8D20 Load=5.0(3.1)
Last=Initialized [2049/1022647632]
== Got message RegisterMessage
Auto logging into 200
-- --* 202
-- --* 201
Auto logging into 199
-- --* 202
-- --* 201
-- --* 200
== Sending Packet Type RegisterAckMessage (24 bytes)
== {SelectSoftKeysMessage} lineInstance=0 callReference=0
softKeySetIndex=0 validKeyMask=126/127
== Sending Packet Type SelectSoftKeysMessage (20 bytes)
== Sending Packet Type DisplayPromptStatusMessage (48 bytes)
== Sending Packet Type CapabilitiesReqMessage (4 bytes)
== Got message IpPortMessage
== Got message HeadsetStatusMessage
== Got message CapabilitiesResMessage
Device has 7 Capabilities
-- CODEC: 4 - G.711 u-law 64k
-- CODEC: 2 - G.711 A-law 64k
-- CODEC: 11 - G.729
-- CODEC: 12 - G.729 Annex A
-- CODEC: 15 - G.729 Annex B
-- CODEC: 16 - G.729 Annex A+Annex B
-- CODEC: 25 - Wideband 256k
== Got message HeadsetStatusMessage
== Got message ButtonTemplateReqMessage
== Configuring button template. buttonOffset=0, buttonCount=20,
totalButtonCount=20
-- 1 9
-- 1 0
-- 1 0
-- 1 0
-- 1 0
-- 1 0
-- 2 9
-- 1 2
-- 2 2
-- 3 2
-- 4 2
-- 5 2
-- 6 2
-- 7 2
-- 8 2
-- 9 2
-- 10 2
-- 11 2
-- 12 2
-- 13 2
== Sending Packet Type ButtonTemplateMessage (100 bytes)
== Got message SoftKeyTemplateReqMessage
== Sending Packet Type SoftKeyTemplateResMessage (656 bytes)
== Got message SoftKeySetReqMessage
-- Set[0] = 0:1 1:2 2:5 3:3 4:9 5:10 6:16 7:17 8:18 9:4 10:14
11:13 --
-- Set[1] = 0:3 1:9 2:4 3:14 4:13 5:19 6:10 --
-- Set[2] = 0:10 1:2 --
-- Set[3] = 0:11 --
-- Set[4] = 0:1 1:9 2:5 3:16 4:17 5:18 --
-- Set[5] = --
-- Set[6] = 0:8 1:9 --
-- Set[7] = --
-- Set[8] = --
-- Set[9] = 0:1 1:9 --
-- Set[10] = 0:21 --
-- There are 11 SoftKeySets.
== Sending Packet Type SoftKeySetResMessage (784 bytes)
== Got message LineStatReqMessage
== Sending Packet Type LineStatMessage (76 bytes)
== Got message LineStatReqMessage
== Sending Packet Type LineStatMessage (76 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 13
-- speeddial 13 not assigned
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 12
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 11
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 10
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 9
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 8
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 7
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 6
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 5
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 4
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 3
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 2
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message SpeedDialStatReqMessage
-- Speed Dial Request for Button 1
== Sending Packet Type SpeedDialStatMessage (72 bytes)
== Got message unknownClientMessage2
== Got message TimeDateReqMessage
== Sending Packet Type DefineTimeDate (40 bytes)
==
Re: [Asterisk-Users] Am I missing something really basichere?????helpwith Asterisk@home {Scanned} {Scanned} {Scanned}
I have no idea what AMP configurator is? http://amp.coalescentsystems.ca/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail attachment not being emailed out {Scanned}
I'm new at this too.
In my voicemail.conf under general I have attach=yes.(This works for all
users) I did try removing it and adding to the end of my users voicemail
entries. Run the test and no attachment. But I'm still new.
David
On Thu, 2005-01-27 at 18:42 -0500, Jeff R Glassman wrote:
I am running [EMAIL PROTECTED]
Voicemail works fine but does not email out the voicemail attachments. Any
suggestion?
---
Voicemail.conf
[general]
#include vm_general.inc
#include vm_email.inc
[default]
201 = {password},Jeff G Laptop,[EMAIL PROTECTED],,attach=yes
-
Sip.Conf
[201]
username=201
type=friend
secret={ACCOUT PASSWORD}
qualify=no
port=5060
nat=yes
mailbox=201
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Jeff G Laptop 201
Jeff
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--
David Shaw [EMAIL PROTECTED]
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Re: [Asterisk-Users] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk
VAD == voice activity detect Steve Alberto Fernandez wrote: VAD, Voice Activity Detection On Thu, 2005-01-27 at 21:16 -0500, Brian Dingman wrote: PCMU is g711 ULAW and PCMA is G711 ALAW. ALAW is more common in Europe. Not sure about VAD. On Thu, 27 Jan 2005 18:44:09 -0700, Kim Lux [EMAIL PROTECTED] wrote: Thanks for the tips. The Grandstream doesn't have a G711 or uLaw option for codecs. It has PCMU, PCMA and iLBC. Are any of these related to G711 ? Grandstreams have echo cancellation and it appears to be working after a few seconds of conversation. What is VAD ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA-841 with Asterisk
Hi, Just received my new SPA-841 phone and I am trying to find a comprehensive how-to with Asterisk without luck. Anyone has that working? Anyone can list high level steps or point me to a how-to somewhere ? Thanks Stephane [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail attachment not being emailed out {Scanned}
I lied it did email me an attachment. Check voice-mail entree line. it
has two comas ,, in it.
I rem out the attach=yes in my voicemail.conf file.
Then added attach=yes at the end of my entree.
101 = {passwd},David,[EMAIL PROTECTED],attach=yes
Works great..
David
On Thu, 2005-01-27 at 18:42 -0500, Jeff R Glassman wrote:
I am running [EMAIL PROTECTED]
Voicemail works fine but does not email out the voicemail attachments. Any
suggestion?
---
Voicemail.conf
[general]
#include vm_general.inc
#include vm_email.inc
[default]
201 = {password},Jeff G Laptop,[EMAIL PROTECTED],,attach=yes
-
Sip.Conf
[201]
username=201
type=friend
secret={ACCOUT PASSWORD}
qualify=no
port=5060
nat=yes
mailbox=201
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Jeff G Laptop 201
Jeff
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--
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[Asterisk-Users] zap FXO channel - wait for N seconds before answer
Is there any way to configure a zap channel to wait for some period of time or number of rings before answering the line? I would like to have a line shared between in-house emergency phones and the asterisk PBX. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] redirect different phone number to different IP phone
On Fri, Jan 28, 2005 at 09:25:55AM -0500, Andrew Thompson said: Video Dery / Internet du Royaume wrote: Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers What kind of line? There has been some questions in the last day or so about DNIS, so I'm not sure that it can be done on inbound analog lines. I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 exten=5551234,1,Dial(SIP/phone1) exten=5551235,1,Dial(SIP/phone2) Customize accordingly... If on analog, you may be able to use distinctive ringing (zapata.conf) There are examples in the config file. Note that I think the dring code is limited to one zap interface which is unfortunate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Authentication against voicemail password database
I would like to allow my remote users to dial in from their homes, cells, etc., and instruct Asterisk to forward calls made to their office extension to a number of their choosing. The wiki entry on Asterisk call forwarding shows how to do this. For security purposes, I would like to front-end this by asking the user to supply a password for their extension. Ideally, this would be their voicemail password. Is there a cmd I can use in extensions.conf to check extension and password against the voicemail database? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where can I find good doc on AGI?
http://home.cogeco.ca/~camstuff/agi.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Friday, January 28, 2005 4:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Where can I find good doc on AGI? Hi, I have searched the list/Wiki, web and I am not able to find a decent documentation of the AGI/FastAGI interface with examples. Am I looking in wrong places? Help will be greatly appreciated. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP + NAT = horrible mess
I don't think you can use NAT = yes unless there is a STUN server involved. See my post yesterday for my Grandstream settings. No, I had nat=yes working with my Cisco 7960 which did not provide it's public IP. However, you need to tell the IP Phone to start using the IP and port that * received the SIP messages from for RTP traffic (use via IP address and via port). -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd and Tollfree
Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + NAT = horrible mess
NAT=yes Rules STUN=SUCKS rtp streams =Rules I have lots of devices connected behind NAT without trouble but in fact with STUN was a real MESS regards Humberto On Fri, 28 Jan 2005 10:40:25 -0500, Nabeel Jafferali [EMAIL PROTECTED] wrote: I don't think you can use NAT = yes unless there is a STUN server involved. See my post yesterday for my Grandstream settings. No, I had nat=yes working with my Cisco 7960 which did not provide it's public IP. However, you need to tell the IP Phone to start using the IP and port that * received the SIP messages from for RTP traffic (use via IP address and via port). -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeelatjafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I want to display my numbers for incoming calls when some one dials my number from any where
On Thu, 2005-01-27 at 23:59 -0800, Mazhar Hussain wrote: Hi to all again, Thanks for your quick response as you siad set callerid via extensions.conf by using apps available. Look at show applications via the asterisk CLI. can i write context for incoming , like Why don't you actually do some work on the problem yourself? http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SetCIDName -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-841 with Asterisk
Stephane Ricard wrote: Hi, Just received my new SPA-841 phone and I am trying to find a comprehensive how-to with Asterisk without luck. Anyone has that working? Anyone can list high level steps or point me to a how-to somewhere ? This assumes that Asterisk and the phone are on the SAME LAN and the phone has ALL FACTORY DEFAULTS and the phone is running 0.9.1 firmware and the phone is using DHCP. You MAY have to set the Go into the web server interface for the phone. Pick Admin Login in the upper right Pick Ext 1 Fill in the Proxy, User ID, and Password fields. Select Submit All Changes Repeat for Ext 2 In sip.conf in Asterisk [theusername] type=friend username=theusername secret=thepassword host=dynamic context=myhappycontext disallow=all allow=ulaw extensions.conf Dial(SIP/theusername) --Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with chan_sccp and cisco 7960
I'm wondering why are you using SCCP and not SIP as most of us that use Cisco 7960 phones? Martin Mostly because 7914 addon module is not supported in SIP images for 7960. Alternative, SIP solution, for a device like 7960+7914 could be Snom 220 + Keypad 220, but I still didn't managed to get it and test it. If anyone knows a good working SIP solution for telephone with keypad with light indications for channel states, please let me know. Nenad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoiding queue retries without hangs?
On Thu, 2005-01-27 at 20:35 +0100, Bruno Hertz wrote:
Anybody found a way around this (bug?), i.e. avoiding retries with
Queue(...|t) properly timing out at the same time ?
OK, I took a look at app_queue.c, and while the described behavior isn't
a bug, I still hacked the source to give me a different retry semantics.
Specifically, if retry=0 the original strategy is to set it to a default
value of 5. My hack is to don't do any retries in this case anyway and
behave the same way as if the call timed out on the queue.
For those interested, the changes to app_queue.c are small:
in reload_queues
- if (q-retry 1)
+ if ( (q-retry 1) (q-retry != 0) )
q-retry = DEFAULT_RETRY;
in queue_exec
/* Leave if we have exceeded our queuetimeout */
if (qe.queuetimeout ( (time(NULL) - qe.start) = qe.queuetimeout) ) {
res = 0;
break;
}
+ if ( (qe.parent)-retry == 0 ) {
+ res = 0;
+ break;
+ }
That's it. That way, no retries are attempted at all if retry=0, and
Queue times out if it does so on the queue members, i.e. according to
timeout in queue.conf. Tested though only with ringall, don't sure how
it works with other ringing strategies.
Thanks, Bruno.
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Re: [Asterisk-Users] zap FXO channel - wait for N seconds before answer
From the asterisk demo: exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,1,Answer ; Answer the line You can also wait 10 sec, 30 sec, etc... to allow as many rings as you like. Cheers, Jon. On Friday 28 January 2005 09:08 am, Steven P. Donegan wrote: Is there any way to configure a zap channel to wait for some period of time or number of rings before answering the line? I would like to have a line shared between in-house emergency phones and the asterisk PBX. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error while trying to install astcc
Hello list, Here is the error im getting when i try to make install with astcc. Can somebody know this error and how to fix it? [EMAIL PROTECTED] astcc]# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi /dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 .) at ./astcc.agi line 47. BEGIN failed--compilation aborted at ./astcc.agi line 47. make: *** [install] Error 2 Regards. image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stumped by BroadVoice SIP
I tried everything and only got that configuration with all bv servers listed to work. -Original Message- From: Luki [mailto:[EMAIL PROTECTED] Sent: Thursday, January 27, 2005 8:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stumped by BroadVoice SIP And need to have all of their four servers to listen to incoming calls as ony one can send it in.. Why do you think that? The server you registered last with will send incoming calls. I've registered my several lines only at their LAX server for the last few months, and didn't miss a call. But even if that's not the case, use permit=147.135.0.0/20 and that would cover their four locations. In /etc/hosts you can select which BV server you want to use. Other than that, my config is pretty much the same as your first broadvoice section, except nat=no. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CISCO 7905 Phone Weirdness
Well, it seems to be acting normal since I wrote the message yesterday. On your thoughts, I haven't messed with ethereal yet, so I am not sure about that. Plus I am not sure if the network item that this phone plugs into is a hub or a switch, I think it is a switch though. I know ethereal won't work if the network item is a switch. On your second suggestion, the 7905 has a web interface, but not a telnet interface, but I will look into and report back. Dan On Fri, 28 Jan 2005, Rich Adamson wrote: It seems on my phone, which is hooked up to a large pbx network powered by an asterisk server, that it will randomly start ringing with a callerid# of 2013 which is its username for that phone. I have looked and been watching on the asterisk command line with the -vvr switch and nothing has been seen that indicates a reason for this random ringing. This leads me to think that this trouble is not involving the asterisk server, however I am bothered by it and would like to find out what is causing the trouble. I was wondering, does anyone here have any ideas as to what might be causing the weird riging, or if not what is causing it, are there any suggestions as to what to look into to find a possible source of the problem. I don't use the 7905's, but might try: - use ethereal to see if the call is actually initiated from something external to the phone - if you are using sip firmware, implement telnet within the phone, telnet to it, and look at some of the debug options ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX outgoing redundancy
So when should you receive a NOANSWER back? Doesn't that imply you are using DIAL with a timeout value? Otherwise I can't see how you would ever get there. I agree with you about LiveVoip. They claim to be an Asterisk service provider but anytime you have a problem they tell you that asterisk is full of bugs and not their only supported platform. On Fri, 28 Jan 2005 02:21:36 -0500, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On January 27, 2005 11:20 pm, Brian Dingman wrote: To combat this problem you will want to change the following line to actually do something: exten = dial-NOANSWER,1,Hangup That's a *large* failure on LiveVoip's part, IMO. If I get a NOANSWER back I don't *want* to do anything -- there was no answer so I don't want to try to dial out again through another provider. I've tried pretty much every VOIP provider out there... nufone (for me) has been the absolute best. I've *never* had any of this bullshit I'm seeing on the list like I am with the Broadvoice and LiveVoip type providers. it just effing works. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stumped by BroadVoice SIP {Scanned}
None just one single line.
-Original Message-
From: David Shaw [mailto:[EMAIL PROTECTED]
Sent: Friday, January 28, 2005 6:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Stumped by BroadVoice SIP {Scanned}
Manjit, Do you have 3 lines with BroadVoice? If so how do you tell which
number is ring in on or which line to dial out on I have on line
with him now and would like to add two lines..
Thanks, David.
On Thu, 2005-01-27 at 14:14 -0800, Manjit Riat wrote:
I had a lot of problem with them to set up..
You need to register to sip.broadvoice.com
And need to have all of their four servers to listen to incoming calls as
ony one can send it in..
Just posted my config two days ago.
http://lists.digium.com/pipermail/asterisk-users/2005-January/085736.html
hope that helps
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 2:02 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Stumped by BroadVoice SIP
Hello guys.
I am a fairly new user to Asterisk, and I'm just having a tough time.
My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
number, and I have three systems with SIP phones.
The PBX and the SIP phones are all behind a Cisco PIX running NAT.
I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
little luck.
The SIP phones are two X-Lites on Windows and one Kphone on Linux (running
from the same system that Asterisk runs on).
It appears that the BroadVoice SIP registers and the SIP phones register,
as I can call from one Xlite to the Kphone. However, I cannot get
incoming calls from BroadVoice. Calling the BroadVoice number results in
a 'The party you wish to reach is busy and cannot...' message. I sniffed
packets and I can see packets coming in from BroadVoice on port 5060 to
the PBX, but they do not correspond with my call attempts. And debugging
the sip session shows alot of '404 Not Found'.
Also, even though this is meant as a incoming only PBX, I tried to test
outgoing calls from an X-Lite softphone to BroadVoice, but it doesn't
work, either.
I've probably screwed my configs to hell trying to get this to work, but
here they are. Any suggestions would be appreciated.
Here are my configs, decrufted...
sip.conf
[general]
context=sip
recordhistory=yes
port = 5060
bindaddr = 0.0.0.0
allow=gsm
allow=alaw
allow=ulaw
allow=adpcm
allow=speex
allow=ilbc
allow=slinear
[general]
nat=yes
register = 212999:password:[EMAIL PROTECTED]:5060
register = 212999:password:[EMAIL PROTECTED]:5060
externip = 208.59.47.2
localnet=192.168.1.0/255.255.0.0
[sip_proxy]
type=user
context=from-broadvoice
[xlite1]
type=friend
regexten=101
username=xlite1
secret=password
callerid=Stephen's Laptop 101
host=dynamic
nat=no
canreinite=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=inband
qualify=yes
[xlite2]
type=friend
regexten=103
context=sip
username=103
secret=password
callerid=Ben's Laptop 103
host=dynamic
nat=no
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=inband
quality=yes
[kphone1]
type=friend
username=kphone1
secret=password
callerid=Diablo 102
host=dynamic
allow=gsm
qualify=yes
[sip.broadvoice.com]
type=peer
host=proxy.dca.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=212999
secret=password
context=incoming
canreinvite=no
[broadvoice-out]
type=peer
dtmfmode=inband
host=147.135.0.128
user=212999
username=212999
authuser=212999
fromuser=212999
fromdomain=sip.broadvoice.com
md5secret=password
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
[broadvoice-out2]
type=peer
dtmfmode=inband
host=147.135.8.128
user=212999
username=212999
authuser=212999
fromuser=212999
fromdomain=sip.broadvoice.com
md5secret=password
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
[broadvoice-incoming]
type=peer
dtmfmode=inband
host=147.135.8.128
context=incoming
qualify=yes
nat=yes
canreinvite=no
fromdomain=sip.broadvoice.com
username=212999
fromuser=212999
insecure=very
[broadvoice-incoming2]
type=peer
dtmfmode=inband
host=147.135.0.128
context=incoming
qualify=yes
nat=yes
canreinvite=no
fromdomain=sip.broadvoice.com
username=212999
fromuser=212999
insecure=very
-
extensions.conf
-
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1; MSD digits to
Re: [Asterisk-Users] Authentication against voicemail password database
Adam Robins wrote: I would like to allow my remote users to dial in from their homes, cells, etc., and instruct Asterisk to forward calls made to their office extension to a number of their choosing. The wiki entry on Asterisk call forwarding shows how to do this. For security purposes, I would like to front-end this by asking the user to supply a password for their extension. Ideally, this would be their voicemail password. Is there a cmd I can use in extensions.conf to check extension and password against the voicemail database? The only thing that comes to mind for me is loading the voicemail configuration from a database, and using an AGI that can read that database to authenticate and process your call forwarding. An upside to this might be the ability to allow users to change their own password(which I'm not sure they can do with voicemail.conf). -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk HEAD - Stable schedule?
Roy Sigurd Karlsbakk wrote: When was this? Sorry, I don't remember when... it may have been on Asterisk Daily News. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.0.3-BRIstuffed
Hi! I have 1.0.3-BRIstuffed and hfc-s card ztcfg says that card is configured (2 B and 1 D channel) but asterisk don't pickup any calls. Any ideas? Regards, Corvin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Minimum Setup
Title: Minimum Setup Hi all, I have asterisk installed and working just fine with a couple of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and am wondering what hardware would be recommended to make minimum connectivity to the public telephone network. I am think ISDN as I would like a few external lines to be accessible. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily consitute an emergency on my part. This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * acting as IP-Phone?
Hi, is it possible, that my * identifies himself as ip-phone? I.e. Im using a grandstrem 100 phone and if I use * as proxy, the authentification string should be changed. Im not sure where looking for this. Hfh, Oliver ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom Phones
Contacted Scott Willard at Polycom this morning, he has since been reassigned to other duties within the organization. Mr. Willard's tone seemed optimistic, and he referrred me to Roger Austin, Regional Channel Manager for Voice. Roger's reply to my inquiry is as follows: Cory, We appreciate your interest in Polycom VoIP Phones. Polycom deploys our VoIP phones with our VoIP Platform partners and at this time those Partners are Sphere, Broadsoft, Sylantro, and Interactive Intelligence. Unfortunately we are not supporting the Asterisk solution at this time. I am going to continue to pursue this, but this is the pushback I have gotten thusfar. We are a Polycom authorized reseller, and have compiled some pretty detailed documentation of workarounds and fixes for typical Polycom/Asterisk integration issues. My engineering folks monitor the forum(s) and I have encouraged them to respond to any developer posts where they feel they can offer some insight or solution. From what I have seen/heard/read at least publically, Asterisk is still not registering on the radar of the larger vendors. I'm curious at what point this might change I guess we'll have to wait and see. Cory Andrews Senior Partner VOIPSupply.com + V: 800.398.VOIP X22 F: 716.630.1548 E: [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: - Original Message - Hmm. Your own web site has it priced between the 500 and 600. If the difference is good support versus zero support, wouldn't the $50 difference between the 500 and the 480i be saved in the first 20 minutes you spend fighting with a problem? Another factor is that one company tests with * and the other shuns it. Just the availability of the firmware alone is almost worth the $50. Just a heads-up for those that use the Sayson 480i phones... Official word from Sayson is that their entire development team for the 480i has been reassigned to develop the firmware for the as yet unreleased 9113i phone. Firmware updates have been deferred for at least 3 months, with the much anticipated XML support being deferred as well. Regerds, Derek Bruce ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with H323/G729--No NATting and no Dynamic IP involved...
Hello... I'm having problems with H323/G729 setup. Below is the output
of h.323 debug when making a call. I use a SIP phone connected to an *
box in the same LAN. The * connects to a h323/g729 PSTN terminator
through internet. Calls rings and are answered in the other side, but I
get no sound at all nor the other side does (complete silence in both
sides). I thought this would just happen when:
1- Codecs conflicts
2- NATting problems
Neither of this circumstances occur now. * have a public IP and no
Firewall nor NAT device. Used codec, as you can see, is G729... my only
concern is that I see G.729A{sw} and G.729{sw} as different codecs in
the Allowed Codecs table and then you see Started logical channel:
receiving G.729{sw} and Started logical channel: sending
*G.729A{sw}... Notice the A in the G729 receiving and the lack
of it in sending.
Might be this subtle difference the cause of my problem?
Thanks in advance for the help.
RODOLFO
---
*CLI h.323 debug
H323 debug enabled
-- Executing Dial(SIP/12345-4cbb,
H323/[EMAIL PROTECTED]) in new stack
Allowed Codecs:
Table:
G.729A{sw} 1
G.729{sw} 2
Set:
0:
0:
G.729A{sw} 1
G.729{sw} 2
-- Making call to [EMAIL PROTECTED]
== New H.323 Connection created.
-- 12345 is calling host [EMAIL PROTECTED]
-- Call token is ip$localhost/26043
-- Call reference is 26043
-- Called [EMAIL PROTECTED]
-- Sending SETUP message
=-= In OnAlerting for call 26043: sessionId=0
--- no logical channels
-- Ringing phone for xx.xx.xx.xx
-- H323/xx.xx.xx.xx is ringing
=*= In CreateRealTimeLogicalChannel for call 26043
-- externalIpAddress: yy.yy.yy.yy
-- externalPort: 18260
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.729{sw}
-- channelsOpen = 1
=*= In CreateRealTimeLogicalChannel for call 26043
-- externalIpAddress: yy.yy.yy.yy
-- externalPort: 18260
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.729A{sw}
-- channelsOpen = 2
=-= In OnConnectionEstablished for call 26043
-- Connection Established with xx.xx.xx.xx
-- H323/xx.xx.xx.xx answered SIP/12345-4cbb
=-= In OnReceivedAckPDU for call 26043
channelsOpen = 1
-- ClearCall: Request to clear call with token ip$localhost/26043
-- Sending RELEASE COMPLETE
== Spawn extension (sip_default, xx, 1) exited non-zero on
'SIP/12345-4cbb'
channelsOpen = 0
-- Call with xx.xx.xx.xxcompleted (EndedByLocalUser)
== H.323 Connection deleted.
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[Asterisk-Users] Eyebeam - asterisk - Messenger
Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --- Wanna buy a duck? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.0.3-BRIstuffed
dzien dobry! raczej dobry wieczór it means good evening :-) what does it say on the console when you start asterisk with asterisk -c -vvv ? that should get you further :) Heh, that is problem I've compilled bristuff, launch make load for TE mode. I also added to modules.conf load = chan_zap.so and nothing happens. I can dial on s extension, with i4l and chan_capi but not with chan_zap. I've hear that this channel is good about echo, so I wan't to try. i4l causes a lot o echo and pure quality, chan_capi better quality but I can't attach more than one msn to card. But to the point when I lauch chan_zap.so and give -cpv asterisk starts up normally without any errors. Cheers, Corvin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reason 24 (Call ended with Q.931 cause)
Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error reason 24 (Call ended with Q.931 cause) I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver -- Version: 0.7.1 Listening on address: 0.0.0.0:1720 Gatekeeper used: No gatekeeper FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported formats in pref. order: g7290 Jitter buffer limits (min/max): 20-500 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: 3 Max number of inbound H.323 calls: 10 Max number of outbound H.323 calls: 10 Max number of simultaneous H.323 calls: 10 Max call rate (ingress direction): 1.00/30 Starting simple switch on 'Zap/3-1' -- Executing Wait(Zap/3-1, 1) in new stack -- Executing Dial(Zap/3-1, OH323/[EMAIL PROTECTED]|10) in new stack -- H.323 call to [EMAIL PROTECTED] with codec(s) g729 Outbound H.323 call 'ip$localhost/263'. -- Called [EMAIL PROTECTED] Call 'ip$localhost/263' cleared. -- H.323 call 'ip$localhost/263' cleared, reason 24 (Call ended with Q.931 cause) Call 'ip$localhost/263' cleared in INIT state. -- OH323/L263 is busy -- Hungup 'OH323/L263' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' Call 'ip$localhost/263' without owner has already been cleared (2). -- Starting simple switch on 'Zap/3-1' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eyebeam - asterisk - Messenger
did you find how to configure video with eyebeam using asterisk because i wasn`t able to do it yet as well i want to se messangin with it ThanK You On Fri, 28 Jan 2005 13:23:46 -0500 (EST), Francois Meehan [EMAIL PROTECTED] wrote: Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --- Wanna buy a duck? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-841 with Asterisk
Eric Wieling aka ManxPower wrote: Stephane Ricard wrote: Hi, Just received my new SPA-841 phone and I am trying to find a comprehensive how-to with Asterisk without luck. Anyone has that working? Anyone can list high level steps or point me to a how-to somewhere ? This assumes that Asterisk and the phone are on the SAME LAN and the phone has ALL FACTORY DEFAULTS and the phone is running 0.9.1 firmware and the phone is using DHCP. You MAY have to set the Go into the web server interface for the phone. Pick Admin Login in the upper right Pick Ext 1 Fill in the Proxy, User ID, and Password fields. Select Submit All Changes Repeat for Ext 2 In sip.conf in Asterisk [theusername] type=friend username=theusername secret=thepassword host=dynamic context=myhappycontext disallow=all allow=ulaw extensions.conf Dial(SIP/theusername) You should also go into the Phone tab and set Line Key 2 / Extension to be 2 instead of the default of 1. If you don't do this you may have problems calling the second line. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.0.3-BRIstuffed
that should get you further :) , so I wan't to try. i4l causes a lot o echo and pure quality ehh, want and poor quality :/ BR, Corvin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanIsAvail not working
Hi!
I'm testing ChanIsAvail context and it is not working for me.
exten = 55,1,ChanIsAvail(SIP/11SIP/21)
exten = 55,2,Cut(theChannel=AVAILCHAN,,1)
exten = 55,3,Dial(${theChannel},r)
exten = 55,4,Hangup
exten = 55,102,Goto(s,4)
According to notes:
The channels are checked in the order listed, returning the first
available channel in the list in ${AVAILCHAN}.
so when my SIP/21 is available, and it is, it should ring it but it is
not.
This is a guess: SIP/11 is not an appropriate channel name. Use show
channels or sip show channels to see the difference.
Cheers, Philipp
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Re: [Asterisk-Users] reason 24 (Call ended with Q.931 cause)
Turn on debug isdn q931 term mon on your 5350. It is an ISDN signalling error. Strange it is showing up in asterisk through a 323 trunk though... What happens when you do a csim start xxx where xx = phone number to dial from the 5300? -Greg Tola Ogunsan wrote: Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error reason 24 (Call ended with Q.931 cause) I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver -- Version: 0.7.1 Listening on address: 0.0.0.0:1720 Gatekeeper used: No gatekeeper FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported formats in pref. order: g7290 Jitter buffer limits (min/max): 20-500 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: 3 Max number of inbound H.323 calls: 10 Max number of outbound H.323 calls: 10 Max number of simultaneous H.323 calls: 10 Max call rate (ingress direction): 1.00/30 Starting simple switch on 'Zap/3-1' -- Executing Wait(Zap/3-1, 1) in new stack -- Executing Dial(Zap/3-1, OH323/[EMAIL PROTECTED]|10) in new stack -- H.323 call to [EMAIL PROTECTED] with codec(s) g729 Outbound H.323 call 'ip$localhost/263'. -- Called [EMAIL PROTECTED] Call 'ip$localhost/263' cleared. -- H.323 call 'ip$localhost/263' cleared, reason 24 (Call ended with Q.931 cause) Call 'ip$localhost/263' cleared in INIT state. -- OH323/L263 is busy -- Hungup 'OH323/L263' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' Call 'ip$localhost/263' without owner has already been cleared (2). -- Starting simple switch on 'Zap/3-1' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk call flow diagrams for ser voicemail combo
Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a timeout message?...I know SER sends a notify message to asterisk at some stage but im not sure of the exact sequence or if asterisk contacts ua1 directly or through ser. Somekind of call flow diagrams for this implementation wold be great. Im also trying to implement this in practice. I have ser as a registrar and asterisk set up aswell. I have modifed ser.cfg to rewritehostport(asterisk ip:5061) when not found, however could someone tell me what to modify in my sip.conf,exntensions,voicemail.conf? A simple example if possible please because all the examples I havee seen so far have pstn forwrading implemented also which complicates things. A look at someones working version of these would be great! All help appreciated, Thank you, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: FAQ missing info? Asterisk@home V 0.4
Just installed V 0.4 of [EMAIL PROTECTED] Programmed up 3 sip budgetone extensions, they call call each other fine. Tried to dial '9' for an outside line through an X100P to a packet8 ATA but got 'all circuits are busy now'. Here is the console output. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-8d25' -- Executing SetGroup(SIP/30-5dde, 30) in new stack -- Executing Dial(SIP/30-5dde, ZAP/g0/19172073420||) in new stack == Everyone is busy/congested at this time -- Executing Macro(SIP/30-5dde, outisbusy) in new stack -- Executing Playback(SIP/30-5dde, allison7/all-circuits-busy-now) in ne w stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/30-5dde, allison7/pls-try-call-later) in new s tack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro(SIP/30-5dde, hangupcall) in new stack -- Executing ResetCDR(SIP/30-5dde, w) in new stack -- Executing NoCDR(SIP/30-5dde, ) in new stack -- Executing Wait(SIP/30-5dde, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/30-5dde' i n macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/30-5dde' in macro 'outisbusy' == Spawn extension (from-internal, 919172073420, 103) exited non-zero on 'SIP/ 30-5dde' -- Executing Macro(SIP/30-5dde, hangupcall) in new stack -- Executing ResetCDR(SIP/30-5dde, w) in new stack -- Executing NoCDR(SIP/30-5dde, ) in new stack -- Executing Wait(SIP/30-5dde, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/30-5dde' i n macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-5dde' asterisk1*CLI any thoughts? Cheers, Dean -Original Message- From: dean collins Sent: Friday, January 28, 2005 10:56 AM To: 'andrew' Subject: RE: FAQ missing info Btw, do you need the pstn line for the X100P plugged in while running the install? Just downloaded and burning the cd now. -Original Message- From: andrew [mailto:[EMAIL PROTECTED] Sent: Friday, January 28, 2005 12:04 AM To: dean collins Subject: Re: FAQ missing info What card do you have? Version 0.4 supports the X100P automatically. --- dean collins [EMAIL PROTECTED] wrote: Message body follows: hi, I might be missing something basic but are you supposed to edit zaptel file or is it supposed to do it automatically? I've posted this question on the asterisk and amp lists but no one seems to be able to answer me on this. Cheers, [EMAIL PROTECTED] -- This message has been sent to you, a registered SourceForge.net user, by another site user, through the SourceForge.net site. This message has been delivered to your SourceForge.net mail alias. You may reply to this message using the Reply feature of your email client, or using the messaging facility of SourceForge.net at: https://sourceforge.net/sendmessage.php?touser=1157926 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] two OpenH323 vulnerabilities
www.sans.org has two vulnerabilities in OpenH323, one as 'high', one as 'other'. Jason Sjobeck ICQ 5579183 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival Jittery (bad udp checksum)
Just installed festival from source and the voice is very jittery and I get this a lot in the asterisk CLI (at least once on every call) NOTICE[3236]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum Maybe the packets are malformed so I get the jittery sound. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where can I find good doc on AGI?
Thanks, I have seen that but this is over 2 years old, does it mean that it is still current? robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vassil Kolarov Sent: Friday, January 28, 2005 10:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Where can I find good doc on AGI? http://home.cogeco.ca/~camstuff/agi.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Friday, January 28, 2005 4:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Where can I find good doc on AGI? Hi, I have searched the list/Wiki, web and I am not able to find a decent documentation of the AGI/FastAGI interface with examples. Am I looking in wrong places? Help will be greatly appreciated. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MoH does not de-attach
Hi We have a fairly simple Asterisk setup for a callcenter: around 10-15 operators running SIP softphones (X-Pro) and an Asterisk box connected to a E1 service using a Digium T100P, and to our legacy PBX (NEC) over a Digium TDM400P FXO interfaces. Everything is working except our Music on Hold after a transfer of an incoming call to our old PBX, that does not de-attach itself from the transfer. For example: 1. Call from customer enters Asterisk through E1 channel 2. Call center operator pickups the call, to find out he needs to xfer it to some extension on the old PBX, lets say ext. 300 3. He puts the customer on hold. At that moment, the customer starts to listen to MoH. 4. He pickups another line on the SIP phone, and calls extension 300 of the older PBX, through one of the FXO interfaces, and announces the call 5. Operator transfer the original call to the second line, and de-attach itself from the call The transfer is always successful, but sometimes (lets say bout 50% of the times), the MusicOnHold does not de-attach itself from the call, so both the customer and ext. 300 on the old PBX CONTINUES to hear the MusicOnHold on the background, not being able to mute it. As you may imagine, this is somewhat annoying. We have tried with different MoH setup: initially using mpg123 player, and now using slimserver, but it keeps happening. Any clue whats going on? Thanks for your advice. Regards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
Hi KPJ, btw, there is a problem with make loadNT (zaphfc) and Kernel 2.4 systems that should be fixed. I hope you already know about this IRQ_NONE issue! the problem is with line 578 in zaphfc.c saturn:/usr/src/bristuff-0.2.0-RC5/zaphfc # make loadNT cc -c zaphfc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -Wall -DMODVERSIONS -include /usr/src/linux/include/linux/modversions.h -DCONFIG_ZAPATA_BRI_DCHANS zaphfc.c: In function `hfc_interrupt': zaphfc.c:578: error: `IRQ_NONE' undeclared (first use in this function) zaphfc.c:578: error: (Each undeclared identifier is reported only once zaphfc.c:578: error: for each function it appears in.) zaphfc.c:578: Warnung: `return' with a value, in function returning void zaphfc.c: In function `hfc_findCards': zaphfc.c:939: Warnung: int format, long unsigned int arg (arg 8) make: *** [zaphfc.o] Fehler 1 best regards Jui Klaus-Peter Junghanns wrote: Hi Mark, please take a look at bristuff 0.2.0-RC5 which uses * 1.0.5: http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC5.tar.gz best regards Klaus Am Freitag, den 28.01.2005, 14:35 +0200 schrieb Mark Elkins: I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been quietly busy with a BRIstuffed patch that works against Asterisk Head? I also notice that I can't seem to re-compile the H323 stuff any more... with this release... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.cc / sixtel are they legitimate?
Been using them for just over a month for all outbound calls. Their customer service is prompt and courteous. I use Voicepulse for inbound and turn around for the average ticket I open is about three days whereas for Sixtel it's often been same day. As for the 800 numbers...I don't have one. But if they're charging $0.30 now for an auto-assigned number then their rates have gone up because I'm pretty sure those used to be free. How do they make their money on those? Well, I would think that since most 800 numbers are now recycled it's fairly likely that you will get many wrong-number calls and if you have an IVR running those calls will start to add up at $0.02/min!! -mark On Jan 27, 2005, at 9:21 AM, Jon Gabrielson wrote: Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated. Thanks, Jon. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP + NAT = horrible mess
Nat=yes with the phone behind a nat box and asterisk on a registered IP works just fine with Cisco, Snom, Xlite and others (I haven't tried many of the others, however). I don't think you can use NAT = yes unless there is a STUN server involved. See my post yesterday for my Grandstream settings. On Fri, 2005-01-28 at 10:28 +0100, Radovan.Mihalik wrote: Hello, I try to connect VoIP phones to Asterisk on private network, And use Asterisk as outbound proxy via his public IP. But the localnet and externip with nat=yes, just is not working, I believe it might only rewrite SIP headers but does not touch The rtp stream. Am I right ? R. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kim Lux Sent: Friday, January 28, 2005 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP + NAT = horrible mess Comments below. On Fri, 2005-01-28 at 08:18 +0800, Leo Ann Boon wrote: Kim Lux wrote: I was expecting to have to port forward too and yet our setup doesn't require it, not on the laptop nor on the wireless router. I think as long as the SIP clients open a port on the NATing device and keep them open so the SIP provider can connect to it, all is well, even if STUN isn't used. I was surprised by how easy it was to NAT the Grandstreams. I had visions of having every device being assigned a static IP and having a fistful of port forwards assigned to them on the router. You're connecting to a SIP provider or just Asterisk? Just a provider right now. I'll tackle asterisk in a few days. Most SIP provider use a far-end NAT traversal device like Jasomi, Acmepacket or Kagoo. The NAT traversal device has the intelligence to figure out the UDP port mapping used by the NAT. SER + nathelper has the effect. I guess ignorance is bliss in this case. For my SER setup, most of the time we can just plug the SIP phone into a router and it will work without any special config. Unfortunately, there're certain firewalls like PIX and MS ISA that will fail. In those cases, your best bet is to do port forwarding or use an outbound proxy. IIRC, Vonage also has the same problem. Thanks for sharing this. It may help some poor soul trying to get his SIP device working in these situations. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: phone rings when I'm using it over VOIP - WHY?
[snip] Do you get a call-waiting beep when you're on the phone with the original party? I think this is it, I can hear the beep so that would explain why my phone rings when I'm using it. [snip] What am I doing wrong? In your SPA-3000 setup screens, go to the Line1 page, and under Supplementary Service Settings, set Call Waiting Serv to No. That should stop it accepting a second call when one is in progress. You may need to look on the User1 page too, and set CW setting to No. These SPA-3000 units have a zillion parameters, so it's easy to miss one! Cheers Tony Thanks Tony, yes I suppose I could disable call waiting on SPA-3000 thank you for suggestion. However, I think there is something else that is bugging me, and it could be potentially a bug but I don't know where. When I dial FedEx tall free number over IAX/FWD my phone1 will ring. When I dial UPS tall free my phone2 will ring as phone1 is busy. I can not explain why??? -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having problems with LiveVoIP?
Regarding those comments below. I am not surprised at the answer and I doubt anyone would be that's taken a look at the * code...it's just not the most elegant thing in the world is it? No point in huffing and puffing about inline comments though when you'd have to find and convince who knows how many contributors to the source to use sound documenting techniques. Sigh. FWIW Voicepulse has ongoing problems with Asterisk as well. Now, I would think that VP would contribute whatever patches they find are necessary for the correct operation of *. And I would only hope that LiveVOIP would do the same otherwise they'll have to go on fixing the bugs again and again and again. As an aside: It's pretty bad corporate policy (from a PR perspective) to take a confrontational stance to one's current and potential customers in a public forum. It sure ain't gonna sell anyone on your service. -mark On Jan 27, 2005, at 12:18 PM, Brian Dingman wrote: From Support: Asterisk is full of bugs and in many cases you fix one thing only to have another show up. We suggested users move to 1.0.3 Our team will look at more things in the software as a part of our ongoing support to clients. We are looking at this version as well as 1.0.3 for some other issues now but, Asterisk is not our only platform. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD and IAX2
Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a 480 - user is not online message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks I'm not online. *CLI iax2 show registry Host UsernamePerceived Refresh State 65.39.205.121:4569xxx xxx.xxx.xxx.xxx:4569 60 Registered It looks like I'm registered though, and I can even call my own number fine. Other can't. :s Any suggestions? Anyone got something similar to work? Thanks, Guills ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: FAQ missing info? Asterisk@home V 0.4
I had to redue the zapata.conf Commented it out ans added, Also changed default Zap g0 to Zap 1 (deleted Zap g0) Where did you find [EMAIL PROTECTED] .4 all I see is 0.3 Jeff [channels] language=en ;context=inbound-analog ;context=default context=from-pstn signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of dean collins Sent: Friday, January 28, 2005 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4 Just installed V 0.4 of [EMAIL PROTECTED] Programmed up 3 sip budgetone extensions, they call call each other fine. Tried to dial '9' for an outside line through an X100P to a packet8 ATA but got 'all circuits are busy now'. Here is the console output. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-8d25' -- Executing SetGroup(SIP/30-5dde, 30) in new stack -- Executing Dial(SIP/30-5dde, ZAP/g0/19172073420||) in new stack == Everyone is busy/congested at this time -- Executing Macro(SIP/30-5dde, outisbusy) in new stack -- Executing Playback(SIP/30-5dde, allison7/all-circuits-busy-now) in ne w stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/30-5dde, allison7/pls-try-call-later) in new s tack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro(SIP/30-5dde, hangupcall) in new stack -- Executing ResetCDR(SIP/30-5dde, w) in new stack -- Executing NoCDR(SIP/30-5dde, ) in new stack -- Executing Wait(SIP/30-5dde, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/30-5dde' i n macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/30-5dde' in macro 'outisbusy' == Spawn extension (from-internal, 919172073420, 103) exited non-zero on 'SIP/ 30-5dde' -- Executing Macro(SIP/30-5dde, hangupcall) in new stack -- Executing ResetCDR(SIP/30-5dde, w) in new stack -- Executing NoCDR(SIP/30-5dde, ) in new stack -- Executing Wait(SIP/30-5dde, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/30-5dde' i n macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-5dde' asterisk1*CLI any thoughts? Cheers, Dean -Original Message- From: dean collins Sent: Friday, January 28, 2005 10:56 AM To: 'andrew' Subject: RE: FAQ missing info Btw, do you need the pstn line for the X100P plugged in while running the install? Just downloaded and burning the cd now. -Original Message- From: andrew [mailto:[EMAIL PROTECTED] Sent: Friday, January 28, 2005 12:04 AM To: dean collins Subject: Re: FAQ missing info What card do you have? Version 0.4 supports the X100P automatically. --- dean collins [EMAIL PROTECTED] wrote: Message body follows: hi, I might be missing something basic but are you supposed to edit zaptel file or is it supposed to do it automatically? I've posted this question on the asterisk and amp lists but no one seems to be able to answer me on this. Cheers, [EMAIL PROTECTED] -- This message has been sent to you, a registered SourceForge.net user, by another site user, through the SourceForge.net site. This message has been delivered to your SourceForge.net mail alias. You may reply to this message using the Reply feature of your email client, or using the messaging facility of SourceForge.net at: https://sourceforge.net/sendmessage.php?touser=1157926 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.cc / sixtel are they legitimate?
On Thu, 27 Jan 2005 08:21:35 -0600, Jon Gabrielson [EMAIL PROTECTED] wrote: Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated. Thanks, Jon. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users They work great. I've used them for months. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: FAQ missing info? Asterisk@home V 0.4
It was released yesterday, I believe. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff R Glassman Sent: Friday, January 28, 2005 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4 I had to redue the zapata.conf Commented it out ans added, Also changed default Zap g0 to Zap 1 (deleted Zap g0) Where did you find [EMAIL PROTECTED] .4 all I see is 0.3 Jeff [channels] language=en ;context=inbound-analog ;context=default context=from-pstn signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of dean collins Sent: Friday, January 28, 2005 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4 Just installed V 0.4 of [EMAIL PROTECTED] Programmed up 3 sip budgetone extensions, they call call each other fine. Tried to dial '9' for an outside line through an X100P to a packet8 ATA but got 'all circuits are busy now'. Here is the console output. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-8d25' -- Executing SetGroup(SIP/30-5dde, 30) in new stack -- Executing Dial(SIP/30-5dde, ZAP/g0/19172073420||) in new stack == Everyone is busy/congested at this time -- Executing Macro(SIP/30-5dde, outisbusy) in new stack -- Executing Playback(SIP/30-5dde, allison7/all-circuits-busy-now) in ne w stack -- Playing 'allison7/all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/30-5dde, allison7/pls-try-call-later) in new s tack -- Playing 'allison7/pls-try-call-later' (language 'en') -- Executing Macro(SIP/30-5dde, hangupcall) in new stack -- Executing ResetCDR(SIP/30-5dde, w) in new stack -- Executing NoCDR(SIP/30-5dde, ) in new stack -- Executing Wait(SIP/30-5dde, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/30-5dde' i n macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/30-5dde' in macro 'outisbusy' == Spawn extension (from-internal, 919172073420, 103) exited non-zero on 'SIP/ 30-5dde' -- Executing Macro(SIP/30-5dde, hangupcall) in new stack -- Executing ResetCDR(SIP/30-5dde, w) in new stack -- Executing NoCDR(SIP/30-5dde, ) in new stack -- Executing Wait(SIP/30-5dde, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/30-5dde' i n macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-5dde' asterisk1*CLI any thoughts? Cheers, Dean -Original Message- From: dean collins Sent: Friday, January 28, 2005 10:56 AM To: 'andrew' Subject: RE: FAQ missing info Btw, do you need the pstn line for the X100P plugged in while running the install? Just downloaded and burning the cd now. -Original Message- From: andrew [mailto:[EMAIL PROTECTED] Sent: Friday, January 28, 2005 12:04 AM To: dean collins Subject: Re: FAQ missing info What card do you have? Version 0.4 supports the X100P automatically. --- dean collins [EMAIL PROTECTED] wrote: Message body follows: hi, I might be missing something basic but are you supposed to edit zaptel file or is it supposed to do it automatically? I've posted this question on the asterisk and amp lists but no one seems to be able to answer me on this. Cheers, [EMAIL PROTECTED] -- This message has been sent to you, a registered SourceForge.net user, by another site user, through the SourceForge.net site. This message has been delivered to your SourceForge.net mail alias. You may reply to this message using the Reply feature of your email client, or using the messaging facility of SourceForge.net at: https://sourceforge.net/sendmessage.php?touser=1157926 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
