[asterisk-users] codian with asterisk voice confrance
Dear all I have video confranceing deivice Codian and i want to intergrate asterisk box with codian so voice confrance is possible with codian users means some users have not codian endpoint so thay call join confranceing with SIP PHONE I have configures asterisk and register codian in asterisk now whn i call from asterisk to codian i got IVR and ask me to inter confrance number when i dial confrance number i got error message invalid number but when i drage and drop that SIP Users in codian console it is working fine but now working with IVR Rgd satish patel - Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Huntgroup with asterisk feature
dear all I there any feature of huntgroup in asterisk means when i call on huntgroup number then any available phone in that group rining is there any feature like this ??? Rgd satish patel - Got a little couch potato? Check out fun summer activities for kids.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box
Hi, I am going to be on the road for the next few days and with the variable delay on the mailing list, I am posting this now, 4 days before the conference. If you haven't yet listened or participated, please consider doing it. We have a great kernel of people at all levels of expertise and ideas and questions can be kicked around immediately (well, there's a few milliseconds lag). This Friday we'll be talking about TDM solutions including ATA that do IAX and SIP without opening the box and installing a card. Your experience in this area would be appreciated. If you sell these solutions come over and pimp them. You can find us here: http://AsteriskUsersConference.org At this site there are three main conference pages, how to listen or participate, a player page for the archived recordings and a page with the extension for a SIP connection to the conference bridge. There are also two links to other pages, a related blog and AsteriskTV which will be getting more and better content and more formats due to the issue of Flash not being compatible with 64-bit systems. I'm working on this now and hope to have that done by mid September. If anyone knows how to convert mp3 to oog on a FreeBSD system, let me know. The video issues are going to be more complicated so if you have suggestions, please post them or email them to me. Thanks to the numerous people who have been supportive of these efforts including Mark Spencer and the guys at Digium. randy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to configure zaptel for incoming call
Hi, I am able to dial through asterisk PBX having TE120P card to E1 card running application. Communication was established successfully Now, I want to do the reverse way out. I am using the following configurations 1)zaptel.conf span=1,1,0,ccs,hdb3,crc4 defaultzone=us bchan=1-15,17-31 dchan=16 2)zapata.conf group=1 signalling=pri_net switchtype=euroisdn context=incoming channel=1-15,17-31 What configuration changes is to be done for landing of call to asterisk PBX when dialled from E1 card running application. I was trying to dial out from E1 card running application with extension number 114 and added the following lines in extensions.conf of asterisk configuration files exten=114,1,Answer but asterisk debugging console is giving the error message -- Extension 's' in context 'channelbank' from '' does not exist. Rejecting call on channel 0/1, span 1 Can anybody tell me how to handle the configuration files for extension number to be called from E1 card running application. Thanx and regards, sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure zaptel for incoming call
Sanchal, You may want to make sure that you have immediate=no set for your E1 channels in zapata.conf. This makes asterisk wait for digits, rather than skipping to the s extension on incoming calls. --TS [EMAIL PROTECTED] 7/30/2007 4:14 AM Hi, I am able to dial through asterisk PBX having TE120P card to E1 card running application. Communication was established successfully Now, I want to do the reverse way out. I am using the following configurations 1)zaptel.conf span=1,1,0,ccs,hdb3,crc4 defaultzone=us bchan=1-15,17-31 dchan=16 2)zapata.conf group=1 signalling=pri_net switchtype=euroisdn context=incoming channel=1-15,17-31 What configuration changes is to be done for landing of call to asterisk PBX when dialled from E1 card running application. I was trying to dial out from E1 card running application with extension number 114 and added the following lines in extensions.conf of asterisk configuration files exten=114,1,Answer but asterisk debugging console is giving the error message -- Extension 's' in context 'channelbank' from '' does not exist. Rejecting call on channel 0/1, span 1 Can anybody tell me how to handle the configuration files for extension number to be called from E1 card running application. Thanx and regards, sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UK ISDN2 / BRI setting
I am running asterisk 1.2 with bristuff 0.3.0 and have the following problem: When I make a call out it fails with a chanunavail message but if I make a call in and then make a call out it is successful. I think this is because BT set the Layer 1 to turn off after a period of time. I need to know how to set the Layer1 / 2 status to call rather than permanent which I think will fix the problem. Neil ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE212 or TE220
Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, I am able to dial through asterisk PBX having TE120P card to E1 card running application. Communication was established successfully Now, I want to do the reverse way out. I am using the following configurations 1)zaptel.conf span=1,1,0,ccs,hdb3,crc4 defaultzone=us bchan=1-15,17-31 dchan=16 2)zapata.conf group=1 signalling=pri_net switchtype=euroisdn context=incoming channel=1-15,17-31 What configuration changes is to be done for landing of call to asterisk PBX when dialled from E1 card running application. I was trying to dial out from E1 card running application with extension number 114 and added the following lines in extensions.conf of asterisk configuration files exten=114,1,Dial(SIP/Phone1,20,tr) but asterisk debugging console is giving the error message -- Extension '114' in context 'channelbank' from '' does notexist. Rejecting call on channel 0/1, span 1 Can anybody tell me how to handle the configuration files for extension number to be called from E1 card running application. Thanx and regards, sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK ISDN2 / BRI setting
On Mon, Jul 30, 2007 at 10:09:32AM +0100, asterisk wrote: I am running asterisk 1.2 with bristuff 0.3.0 and have the following problem: Which version of bristuff do you have exactly? asterisk -rx 'zap show version' When I make a call out it fails with a chanunavail message but if I make a call in and then make a call out it is successful. I think this is because BT set the Layer 1 to turn off after a period of time. I need to know how to set the Layer1 / 2 status to call rather than permanent which I think will fix the problem. Should work, IIRC: the driver should ask the chip to wake up the link when you want to call. No special higher-level operation should be required on your side. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
Hi, I'm brazilian. By the way, Why such a question? See you. Ronaldo. Jozeph Brasil wrote: Some brazilian here on list? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lightweight IAX balancer
Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on each machine. It does balancing not per IAX connection, but per call - rewriting call numbers and keeping track of connection status. It's going to be optimized for speed - doesn't do any other modification or audiostream translation - only message passing. If someone's interested -- code + short doc is available at http://www.gradwell.com/tmp/iax_proxy.tar.gz Development will continue - any opinions / comments / contributions are appreciated. Stanisław Pitucha Gradwell Dot Com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel channel reservation
Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel = 1-15 channel = 17-21 ; outgoing group = 2 signalling=pri_cpe channel = 22-28 ; emergency group = 3 signalling=pri_cpe channel = 29-31 How can I avoid that incoming calls are going to the channels 22-31? Regards, Jack ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outbound caller ID
Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use 1 channel from TE110P for data transmission
Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any tutorial? Probably someone around the world as already done this before. Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
Yep! From São Paulo - SP Where we can help? Regards Josué 2007/7/30, Ronaldo [EMAIL PROTECTED]: Hi, I'm brazilian. By the way, Why such a question? See you. Ronaldo. Jozeph Brasil wrote: Some brazilian here on list? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK ISDN2 / BRI setting
0.3.0-pre-1s After working with traditional pabx's in the past I have known the setting of layer 1 to call has fixed this problem. Thanks Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 30 July 2007 11:34 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] UK ISDN2 / BRI setting On Mon, Jul 30, 2007 at 10:09:32AM +0100, asterisk wrote: I am running asterisk 1.2 with bristuff 0.3.0 and have the following problem: Which version of bristuff do you have exactly? asterisk -rx 'zap show version' When I make a call out it fails with a chanunavail message but if I make a call in and then make a call out it is successful. I think this is because BT set the Layer 1 to turn off after a period of time. I need to know how to set the Layer1 / 2 status to call rather than permanent which I think will fix the problem. Should work, IIRC: the driver should ask the chip to wake up the link when you want to call. No special higher-level operation should be required on your side. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Huntgroup with asterisk feature
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 satish patel wrote: dear all I there any feature of huntgroup in asterisk means when i call on huntgroup number then any available phone in that group rining is there any feature like this ??? You can use queues for this purpose. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iD8DBQFGreCMCFu3bIiwtTARAi07AJ0cfwpibikd8eYhaWJ+yGTFzHS2iwCggIm6 PaVPjhn8uxsLuXatKxCYIII= =nAcy -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lightweight IAX balancer
- Tzafrir Cohen [EMAIL PROTECTED] wrote: Interesting. One thing thoough: what's the license of your code? It's MIT - I forgot to add that. I'll stick the banners to files soon, with next update to the package. (along with some fixes, etc) Stanisław Pitucha Gradwell Dot Com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 licenses installed - voicemail has no audio...
I got my G729 licenses installed.I can make calls out and receive calls and the system shows the licenses are in use, however, if I try to call voicemail.. the CLI shows the files are playing, however I don't hear anything. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Locking a device to a codec
You sure about that? Having a config that looks like this: port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=g726 context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown pedantic=no progressinband=no And then a user that looks like this: [570601] username=570601 accountcode=75415 type=friend secret=6edfa qualify=yes port=5060 pickupgroup= nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all allow=g729 context=from-internal canreinvite=no callgroup= callerid=Test VoIP Accounts 570601 Seemed to lock EVERYONE to using g729!!! On 7/27/07, Jaswinder Singh [EMAIL PROTECTED] wrote: in ur sip.conf under the device definition you can set it for example device name is asterisk is pap2 [pap2] username=pap2 secret=blabla type=friend disallow=all allow=g729 Then asterisk will only use g729 for incoming as well as outgoing calls on this device . On 27/07/07, Matt [EMAIL PROTECTED] wrote: Right.. what I'm asking is: If I set my PAP2T to use G723 or G729 outgoing calls from that device go in that format. However, incoming calls to the device from asterisk are running at G711u. The PBX will access any format G711u, G723, G729 or GSM. What do I need to do to make asterisk use the same codec back to the ATA as it is using to the PBX? On 7/27/07, dave cantera [EMAIL PROTECTED] wrote: baji, mhoppes, remember, if you have Only the g729 codec allowed or if this is the only allow= entry in the sip.conf file, callers requesting any other codec will be rejected daveC Baji Panchumarti wrote: On 7/27/07, Matt [EMAIL PROTECTED] wrote: Can someone comfirm my logic here? If I want a phone to use G729 I can set it to use G729... do I also need to set it in Asterisk? I'm thinking no... as long as asterisk WILL do G729... if that's all the device accepts it should go to that codec, yes? (based on my understanding, take it for what it is worth) if allow=all or allow=g729 is in your asterisk configuration (sip.conf / iax.conf ) then asterisk will stream packets in g729 (assuming you have any licesnses needed in place). -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with logged in agents that are not reachable
On 7/30/07, voiplist [EMAIL PROTECTED] wrote:
I noticed that if I have an agent logged in using AgentCallBackLogin
and that agent is unreachable for some reason (SIP phone unplugged)
calls to him/her will completely yack.
For example:
1-Agent 500 is the only one logged into queue number 1.
2-A call comes into queue number 1
3-The call is pushed to agent 500 at extension 21 which is unreachable
because the ethernet cable is unplugged to extension 21's handset.
4-The caller gets hungup on entirely instead of the call going to
another agent or leaving the caller in the queue
I don't expect this to happen but I want to be sure all bases are
covered on light days during shift changes etc.
This is either a problem with your dialplan or your queue configuration.
If you always want your callers to enqueue regardless of agent status, make
sure that joinempty=yes and leavewhenempty=no in queues.conf for that
queue. You may also want to add a
exten = whatever,n,NoOp(${QUEUESTATUS})
right after your call to Queue() to see why the calls are leaving the
dialplan. I suspect that you've got one or the other of those settings not
set properly, so when there are no available agents, your calls exit the
Queue() application with $QUEUESTATUS set to JOINEMPTY or LEAVEEMPTY,
but you don't have anything in your dialplan following Queue(), so they run
off the end of the extension and * hangs up on them.
Note that there is a problem with 1.4.9 that breaks
joinempty=yes/leavewhenempty=no - there's a patch offered to my bug report (
http://bugs.digium.com/view.php?id=10320), but due to other strange
instability observed in 1.4.8 and 1.4.9, I'm back on 1.47.1, so I haven't
tested it yet.
--
j.
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Re: [asterisk-users] TE212 or TE220
On Mon, 2007-07-30 at 02:40 -0700, fateme fatah wrote: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. There are two major differences between the TE212P and the TE220 cards. The first is the connector. The TE212P card will only fit in a 3.3 volt PCI slot, while the TE220 is designed for a PCI Express slot. The second major difference between the cards is echo cancellation. The TE212P comes with an echo cancellation module installed, while the TE220 card comes without one. (You can always add a VPMOCT064 module to it, but it doesn't come bundled with the card.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] announcement server
Hi All, I would like to build a little announcement server with asterisk. Is it possible to do the following: - when * gets the INVITE message, it should send 183 Session in progress back - it should play an announcement message in early media - then, redirect the client to a specified URI (with 303 Moved temporarily) Is is possible ? Can you give me a little example how to solve it with with asterisk (if its possible) ? Thanks, Mitya ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is SIP conntrack status ?
Hi, Reading from various Netfilter mailing lists, I'm wondering whether or not, has anyone ever got a successful experience with SIP conntrack and Asterisk. For instance, this feature was : - introduced in Linux kernel 2.6.16, - improved in 2.6.18 - enhanced in 2.6.22 - I even read something concerning 2.6.23 As I doubt many production systems run 2.6.23, what is this feature status ? Is it better to simply open RTP ports than playing with connection tracking ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Description for each sound files
Hello all, Where can I find a list of description for each sound files provided by the asterisk-sounds-main Debian package? You can find the contents of my /usr/share/asterisk/sounds/ directory at http://paste.debian.net/33679. Thank you in advance. GNUbie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connections broken
On Sun, 2007-07-29 at 14:51 +0100, Thomas Kenyon wrote: iptables -A PREROUTING -t nat -p tcp -i eth0 --dport 4569 -j DNAT --to ip-of-asterisk-box:4569 should work, assuming you have the relevant parts compiled in. Just for your information, IAX traffic is UDP, not TCP. I just thought I'd bring that up so that someone didn't mistakenly open up their firewall for TCP traffic instead of UDP traffic and wonder why IAX traffic wasn't making it through. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI and exec Playback
Hello, I'm looking for a way to play sound file, and control the playback trough web interface. Is it possible to use AGI to play a sound file and then by receiving some event stop playing it, and play another file. The catch is that i want to seek to 1st minute, 5th minute, etc - so regular ControlPlayback with intervals wouldn't fit - i have to use sox to create different file and then jump to it. Also - i have read that in asterisk 1.4. there is SendDTMF trough AMI - is it possible to use that for ControlPlayback? Here i would want regular Forward/Backward buttons on web :) Regards, Atis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Creating an SIP softphone
Hello, I have been reading up on the capabilities of the Asterisk-Java library. I believe that this library can act as an interface between a Java GUI(custom softphone) and the Asterisk server. Seems like the Live API would be easiest to use to make the connection to the Asterisk server and to set-up calls. One thing I am not sure about is how to transmit the audio streams between users' PC's once the calls are routed. I can see that the Asterisk-Java library can't be used for transmitting real-time audio(phone conversations). Would anyone have an idea about how to complete the application I am trying to make. To be clear, I am creating a custom softphone, but can't find much information on how to create the audio transmission. Could anyone provide me with some advice on how to complete this type of softphone. I noticed that there is a JAIN(SIP) API that can be used with java, but I would need more information or examples on how this can be used for my application to use this API. I would prefer to use the SIP protocol, since it seems like its the most common. Thanks in advance, Denis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
Isso nao vai parar? On 7/30/07, Josué Conti [EMAIL PROTECTED] wrote: Yep! From São Paulo - SP Where we can help? Regards Josué 2007/7/30, Ronaldo [EMAIL PROTECTED]: Hi, I'm brazilian. By the way, Why such a question? See you. Ronaldo. Jozeph Brasil wrote: Some brazilian here on list? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Description for each sound files
On Mon, 2007-07-30 at 21:45 +0800, GNUbie wrote: Where can I find a list of description for each sound files provided by the asterisk-sounds-main Debian package? The file core-sounds-en.txt should contain the text of each of the sound files. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
Chineese now in asterisk mailing list? Ary Junior a crit: Isso nao vai parar? On 7/30/07, Josu Conti [EMAIL PROTECTED] wrote: Yep! From So Paulo - SP Where we can help? Regards Josu 2007/7/30, Ronaldo [EMAIL PROTECTED]: Hi, I'm brazilian. By the way, Why such a question? See you. Ronaldo. Jozeph Brasil wrote: Some brazilian here on list? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue stats
As a different approach, QueueMetrics includes a perl script that does the real-time uploading of queue_log data into a database. It is being used in a large number of high load installations worldwide, so I'd say it's a pretty proven solutions, and it's very lightweight. As an added bonus, it is able to upload multiple queue_log files into the same table, will do an auto-sync in case it is interrupted so that you can load data reliably in the case of a crash, will post heartbeat information and can even do some queue_log rewriting on-the-fly. You can find it at: http://queuemetrics.com/download/qloaderd-1.7.tar.gz - full instructions are enclosed in the tar package. It is free as in beer. I hope this helps l. On Sun, 29 Jul 2007 05:11:12 +0200, Anthony Francis [EMAIL PROTECTED] wrote: I am submitting a patch to the Bug tracker next week that will have a manager event fired alongside every queue log write. You can then send the queue information to the database in realtime if you have a manager interface script. If anyone is willing to test this patch once posted, I would appreciate it. Anthony -- Original Message -- From: Philipp Kempgen [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Sat, 28 Jul 2007 12:13:41 +0200 Jay Moore wrote (received 2007-07-28): My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. [message truncated] -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK ISDN2 / BRI setting
On Mon, Jul 30, 2007 at 01:48:12PM +0100, asterisk wrote: 0.3.0-pre-1s After working with traditional pabx's in the past I have known the setting of layer 1 to call has fixed this problem. There have been quite a few updates to briistuff since. and if all of this doesn't help, maybe try uncommenting the line LAYER2ALWAYSUP libpri option of any use in such a case? Or is it for the NT side? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Huntgroup with asterisk feature
can you explain me how it will work caz i have not much idea about asterisk i am beginner so can u explain me how to use queue and how to forward my call to huntgroup Barry L. Kline [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 satish patel wrote: dear all I there any feature of huntgroup in asterisk means when i call on huntgroup number then any available phone in that group rining is there any feature like this ??? You can use queues for this purpose. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iD8DBQFGreCMCFu3bIiwtTARAi07AJ0cfwpibikd8eYhaWJ+yGTFzHS2iwCggIm6 PaVPjhn8uxsLuXatKxCYIII= =nAcy -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating an SIP softphone
JMF ( http://java.sun.com/products/java-media/jmf/ ) for audio... a good example to use JAIN SIP and JMF is the SIP Communicator source code ( https://sip-communicator.dev.java.net/ ) ... []s Ary Junior On 7/30/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello, I have been reading up on the capabilities of the Asterisk-Java library. I believe that this library can act as an interface between a Java GUI(custom softphone) and the Asterisk server. Seems like the Live API would be easiest to use to make the connection to the Asterisk server and to set-up calls. One thing I am not sure about is how to transmit the audio streams between users' PC's once the calls are routed. I can see that the Asterisk-Java library can't be used for transmitting real-time audio(phone conversations). Would anyone have an idea about how to complete the application I am trying to make. To be clear, I am creating a custom softphone, but can't find much information on how to create the audio transmission. Could anyone provide me with some advice on how to complete this type of softphone. I noticed that there is a JAIN(SIP) API that can be used with java, but I would need more information or examples on how this can be used for my application to use this API. I would prefer to use the SIP protocol, since it seems like its the most common. Thanks in advance, Denis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Description for each sound files
On Mon, Jul 30, 2007 at 09:45:10PM +0800, GNUbie wrote: Hello all, Where can I find a list of description for each sound files provided by the asterisk-sounds-main Debian package? You can find the contents of my /usr/share/asterisk/sounds/ directory at http://paste.debian.net/33679. It's in /usr/share/doc/asterisk-sounds-main/sounds.txt.gz , as you should have expected (documentation for package foo normally resides at /usr/share/doc/foo/ and text files that are long enough are gzipped). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Description for each sound files
GNUbie wrote: Hello all, Where can I find a list of description for each sound files provided by the asterisk-sounds-main Debian package? You can find the contents of my /usr/share/asterisk/sounds/ directory at http://paste.debian.net/33679. You would have to contact the person that built the Debian package. The standard Asterisk source code has a list of the the sound files and what the text of each file is in the sounds.txt file. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Huntgroup with asterisk feature
Yes. Use the group= setting in zapata.conf. group=1 then Dial(Zap/g1/5551212) satish patel wrote: dear all I there any feature of huntgroup in asterisk means when i call on huntgroup number then any available phone in that group rining is there any feature like this ??? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use 1 channel from TE110P for data transmission
If your provides has not provisioned any channels on your t1 as data then this wont work. im guessing that for wha you want an FXS post will do On 7/30/07, Marco Mouta [EMAIL PROTECTED] wrote: Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any tutorial? Probably someone around the world as already done this before. Best regards, Marco Mouta -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lightweight IAX balancer
On Mon, 2007-07-30 at 07:01 -0500, [EMAIL PROTECTED] wrote: Date: Mon, 30 Jul 2007 12:19:13 +0100 (BST) From: Stanis?aw Pitucha [EMAIL PROTECTED] Subject: [asterisk-users] Lightweight IAX balancer To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8 Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on each machine. It does balancing not per IAX connection, but per call - rewriting call numbers and keeping track of connection status. It's going to be optimized for speed - doesn't do any other modification or audiostream translation - only message passing. If someone's interested -- code + short doc is available at http://www.gradwell.com/tmp/iax_proxy.tar.gz Development will continue - any opinions / comments / contributions are appreciated. That SW looks like a valuable service. What are the chances you could code it into a module for OpenSER, so OpenSER could deliver both SIP and IAX routing/proxying, without having to rewrite all common parts of OpenSER to deliver its services to SIP? Also, OpenSER/IAX would make calls with mixed IAX/SIP legs easier to manage. And there's probably lots of performance optimization - not to mention deployment optimization. Stanis?aw Pitucha Gradwell Dot Com -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel channel reservation
Jack wrote: Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel = 1-15 channel = 17-21 ; outgoing group = 2 signalling=pri_cpe channel = 22-28 ; emergency group = 3 signalling=pri_cpe channel = 29-31 How can I avoid that incoming calls are going to the channels 22-31? You must contact your telco. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound caller ID
1 No 2 I dont know. 3 Currently in the us the answer is yes On 7/30/07, Vieri [EMAIL PROTECTED] wrote: Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange ISDN Troubles
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ahoy
I'm trying to setup Asterisk on debian etch (with the debian packages)
with a Fritz!Card PCI ISDN card and chan_capi.
Everything seems to be configured the right way (excerpts below),
Asterisk seems to see the ISDN-card but if i try to place a test-call
from the outside i don't see anything on the asterisk-console (set debug
and verbose to 99). Shouldn't i see _something_ on the console, even if
the DID which is dialed isn't configured yet?
I tested the cable and the NTBA both with a ISDN-phone. This worked
properly. But if i plug the cable into the asterisk-server i just get a
number busy-signal and don't see any messages on the asterisk console
or in the log files.
regards
Florian Arthofer
=exerpts=
capiinfo:
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-07 (49.23)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
gertrud*CLI capi show channels
CAPI B-channel information:
Line-Name NTmode state i/o bproto isdnstate ton number
-
ISDN1#02 no- - trans 0x00 ''-''
ISDN1#01 no- - trans 0x00 ''-''
capi.conf:
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de ;set default language
;ulaw=yes;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;Use one interface section for each isdn port!
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
controller=1 ;capi controller number of this interface/port
group=1 ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended
for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;PBX accountcode to use in CDRs
context=isdn-in ;context for incoming calls
echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
echocancel=yes
;echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
devices=2;number of concurrent calls (b-channels) on this controller
;(2 makes sense for single BRI, 30/23 for PRI/T1)
=excerpts-end
- --
Florian Arthofer
Technik Web- und Mailservices/Administrator Web- and Mailservices
lagis Internet Serviceprovider GmbH
Wiener Straße 151, 4021 Linz, Austria
Phone +43(0)732/3400-5636
Fax +43(0)732/3400-5644
E-Mail [EMAIL PROTECTED]
URL http://www.lagis.at
FN 270805 w des Landesgerichtes Linz
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[asterisk-users] iax2 trunk registration with auth rsa
hi all,
I am trunking via iax2 2 asterisk serverses
if both of them have static ip addresses, I can connect them using no
password, password or auth rsa with a pair of keys.
If one of them has dynamic ip address and need to register on the other
server, I can connect them with no password, but I am not able to do that
using keys.
The question is: which is the right register syntax to use when using keys
pair ?
I tried:
register = serverb:[EMAIL PROTECTED]
servera is the name of the public key of servera, seen on serverb (remember
that if I use static ip I can dial from serverb to servera: in my
extensions.conf I have:
exten = _5XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:0},30,t)
here I don't need to specify the key, becouse I dial servera which is an
entry (room) in my iax.conf where is specified inkey and outkey
but in register command it seems no possible to specify other then ip
address or full name
thanks in advance,
Andrea
Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.
Visitate il sito http://www.frameweb.it
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Re: [asterisk-users] Zaptel channel reservation
On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote: Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel = 1-15 channel = 17-21 ; outgoing group = 2 context = hangup-calls signalling=pri_cpe channel = 22-28 ; emergency group = 3 ; keeping your convention and writing the directive explicitly, ; although it is kept implicitly from previous channel: context = hangup-calls signalling=pri_cpe channel = 29-31 and then in extensions.conf: [hangup-calls] ; not sure that this is precisly the right thing to do: exten = s,1,Hangup -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange ISDN Troubles
On Mon, 2007-07-30 at 16:46 +0200, Florian Arthofer wrote: Shouldn't i see _something_ on the console, even if the DID which is dialed isn't configured yet? Unfortunately, I don't think so. You might want to add a pattern match to your dialplan that would match any DID, and see if that helps. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound caller ID
On Mon, Jul 30, 2007 at 10:40:57AM -0400, C F wrote: On 7/30/07, Vieri [EMAIL PROTECTED] wrote: I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) 1 No 2 I dont know. 3 Currently in the us the answer is yes CNID, administratively, is assigned by the originating class-5 end office of the LEC or CLEC. Some carriers will permit you to specify what it should be, administratively, and some switches will accept what you send (definitely over a PRI, definitely at least some generics on a DMS-100, possibly on a BRI, definitely not on some other switches and generics). Whether a VoIP provider will permit you to set it is probably implementation-defined. The FCC has a finger in this pie as well, I believe; there was recent rulemaking about CNID spoofing, the results of which (I *think*) were to impose as a regulation the perfectly sensible limitation that you should only be permitted to send as originating CNID for the end office to propagate Directory Numbers which are administratively yours. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone -- Asterisk -- Linksys SPA3102 -- PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone -- Asterisk-- Linksys SPA3102 -- PSTN (In Use) -- Use IAX Can any one help me with some dial plan logic for this; I'm confused as to the best way around this? Thanks in advance Chris -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble getting sound from a call
Having some issues with getting sound from a call.
I have 4 systems. 3 main systems which handle calls for our 3 locations.
The 4th system is the central voice mail system. When an inbound call
gets passed to someones voice mail its done with an IAX2 connection. The
same happens after hours when we have our night mode set. If you dial
the main number after hours you are passed straight to the voice mail
server where I have an IVR set to answer/handle the calls:
[ivr-1]
include = heading-out
exten = h,1,Hangup
exten = s,1,Set(LOOPCOUNT=0)
exten = s,n,Set(__DIR-CONTEXT=default)
exten = s,n,Set(_IVR_CONTEXT_${CONTEXT}=${IVR_CONTEXT})
exten = s,n,Set(_IVR_CONTEXT=${CONTEXT})
exten = s,n,GotoIf($[${CDR(disposition)} = ANSWERED]?begin)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=60)
exten = s,n,Background(custom/mhi-main-greeting)
exten = s,n,WaitExten()
exten = #,1,Goto(app-directory,#,1)
exten = #,n,dbDel(${BLKVM_OVERRIDE})
exten = #,n,Set(__NODEST=)
exten = #,n,Goto(app-pbdirectory,pbdirectory,1)
exten = hang,1,Playback(vm-goodbye)
exten = hang,n,Hangup
exten = i,1,dbDel(${BLKVM_OVERRIDE})
exten = i,n,Set(__NODEST=)
exten = i,n,Goto(ivr-1,s,begin)
exten = t,1,dbDel(${BLKVM_OVERRIDE})
exten = t,n,Set(__NODEST=)
exten = t,n,Goto(app-blackhole,hangup,1)
exten = 0,1,Goto(incoming,252,1)
[heading-out]
include = call-sa-users
include = call-dal-users
include = call-hou-users
[call-dal-users]
exten = 101,1,Dial(IAX2/toPBX2/${EXTEN})
exten = 101,n,Hangup
exten = 102,1,Dial(IAX2/toPBX2/${EXTEN})
exten = 102,n,Hangup
exten = 103,1,Dial(IAX2/toPBX2/${EXTEN})
exten = 103,n,Hangup
exten = 104,1,Dial(IAX2/toPBX2/${EXTEN})
exten = 104,n,Hangup
[call-hou-users]
exten = 150,1,Dial(IAX2/toPBX3/${EXTEN})
exten = 150,n,Hangup
exten = 151,1,Dial(IAX2/toPBX3/${EXTEN})
exten = 151,n,Hangup
exten = 152,1,Dial(IAX2/toPBX3/${EXTEN})
exten = 152,n,Hangup
exten = 153,1,Dial(IAX2/toPBX3/${EXTEN})
exten = 153,n,Hangup
[call-sa-users]
exten = 200,1,Dial(IAX2/toPBX1/${EXTEN})
exten = 200,n,Hangup
exten = 201,1,Dial(IAX2/toPBX1/${EXTEN})
exten = 201,n,Hangup
exten = 202,1,Dial(IAX2/toPBX1/${EXTEN})
exten = 202,n,Hangup
exten = 203,1,Dial(IAX2/toPBX1/${EXTEN})
exten = 203,n,Hangup
[app-directory]
include = app-directory-custom
exten = #,1,Answer
exten = #,n,Wait(1)
exten =
#,n,AGI(directory,${DIR-CONTEXT},heading-out,${DIRECTORY:0:1}${DIRECTORY_OPTS})
exten = #,n,Playback(vm-goodbye)
exten = #,n,Hangup
exten = i,1,Playback(privacy-incorrect)
If you know the persons extension who you want to call you can dial it
and if they don't answer you get passed back to the voice mail system
and the persons message is played, you can hear it play, and you are
able to leave them a message. The problem comes if you hit # to enter
the directory. Once you find the person you are looking for and you hit
1 to dial them their phone rings, if they pick up you can talk to them
fine and there are no audio problems. If they don't answer and you get
passed back to the voice mail system I see the system answer the call
-- Executing Goto(IAX2/sapeer-1, ivr-1|s|1) in new stack
-- Goto (ivr-1,s,1)
-- Executing Set(IAX2/sapeer-1, LOOPCOUNT=0) in new stack
-- Executing Set(IAX2/sapeer-1, __DIR-CONTEXT=default) in new stack
-- Executing Set(IAX2/sapeer-1, _IVR_CONTEXT_ivr-1=) in new stack
-- Executing Set(IAX2/sapeer-1, _IVR_CONTEXT=ivr-1) in new stack
-- Executing GotoIf(IAX2/sapeer-1, 0?begin) in new stack
-- Executing Answer(IAX2/sapeer-1, ) in new stack
-- Executing Wait(IAX2/sapeer-1, 1) in new stack
-- Executing Set(IAX2/sapeer-1, TIMEOUT(digit)=3) in new stack
-- Digit timeout set to 3
-- Executing Set(IAX2/sapeer-1, TIMEOUT(response)=60) in new stack
-- Response timeout set to 60
-- Executing BackGround(IAX2/sapeer-1,
custom/mhi-main-greeting) in new stack
-- Playing 'custom/mhi-main-greeting' (language 'en')
== CDR updated on IAX2/sapeer-1
-- Executing Goto(IAX2/sapeer-1, app-directory|#|1) in new stack
-- Goto (app-directory,#,1)
-- Executing Answer(IAX2/sapeer-1, ) in new stack
-- Executing Wait(IAX2/sapeer-1, 1) in new stack
-- Executing AGI(IAX2/sapeer-1, directory|default|heading-out|)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/directory
-- Playing 'dir-intro-fn' (language 'en')
== directory|default|heading-out|: Found
/var/spool/asterisk/voicemail/default/231/greet.wav
directory|default|heading-out|: -- Playing 'dir-instr' (language 'en')
-- AGI Script directory completed, returning 0
-- Executing Dial(IAX2/sapeer-1, IAX2/toPBX1/231) in new stack
-- Called toPBX1/231
-- Call accepted by 192.168.81.2 (format ulaw)
-- Format for call is ulaw
-- IAX2/toPBX1-2 is ringing
-- IAX2/toPBX1-2 stopped sounds
-- Accepting AUTHENTICATED call from 192.168.81.2:
requested format = ulaw,
Re: [asterisk-users] G729 licenses installed - voicemail has no audio...
I got my G729 licenses installed.I can make calls out and receive Make sure you add g729 to the voicemail config as well. John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 licenses installed - voicemail has no audio...
Make sure you add g729 to the voicemail config as well. ?? Don't understand. I still want my format=wav|gsm.But that doesn't seem to be the issue... as I can't even hear the password prompts. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound caller ID
On Mon, 30 Jul 2007 05:24:31 -0700 (PDT), Vieri wrote: Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) Nope 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Yepp, as long as the telco allows. In for example sweden, you can only set the CID to the numbers that you own. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Silly MeetMe() question.
I've got the ztdummy kernel module loaded and seem to have all the desired prerequisites in place, but Asterisk never seems to compile with MeetMe() application support enabled, nor does there appear to be a module I am failing to load that would contain this application. Is there something really obvious I am missing? Thanks, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connections broken
On 7/30/07, Jared Smith wrote: Just for your information, IAX traffic is UDP, not TCP. I just thought I'd bring that up so that someone didn't mistakenly open up their firewall for TCP traffic instead of UDP traffic and wonder why IAX traffic wasn't making it through. Amen ! I had changed my router, the calls via my DID were working fine, but I just COULD NOT get either of my soft phones to connect. I looked at the contexts, nothing. The * console was not dead as ever. I check the port forwarding and Bingo ! only TCP was being forwarded. Aaaah ! -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zombie (Masqueraded) Channel CDR Problem
Hi, We are running asterisk 1.2.16 and need to connect two channels which are already established. We are currently using app_meetme to achieve that, but we are sometimes unhappy, as app_meetme provides functionality that produces load that we do not need in our two party conferences. I figured out that there is an alternative called app_changrab. (http://www.freeswitch.org/asterisk_stuff/app_changrab.c) First tests had shown that app_changrab worked well despite CDR logging. Changrab uses mainly (I have very little understanding of the asterisk internals therefore these are just assumptions...) ast_bridge_call to connect the channels. But it doesnt connect the actual channel, but creates a masqueraded (Zombie?) channel that is handed to the ast_bridge_call command. I have seen in the manager interface that the Zombie Channel had the same 'uniqueid' and is hungup instantly after bridging the call. This leads to an undesired behavior: the CDR Engine assumes that the call has ended when the Hungup Zombie channel event has to be handled. The 'duration' and 'billsec' fields (which are the most important ones for our accounting!) show a duration that end with the hungup zombie channel. Is there any workaround? Can I rearrange app_changrab to use the actual channel or fork the CDR in the dialplan (my odbc postgres hates that bcs. of a unique constraint on uniqueid that I can remove) or do some other magic to get the right billsec which is a must for our application? Knud -- Knud A. Müller ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/Dont know how to handle Accepted
El Sun, Jul 29 de 2007 a las 20:04 +0800, Steve Underwood comentaba: What versions of software did you use to get a screwed up result like that? The message Don't know how to handle signalling event Accepted is printed at the end of a case statement which does handle that event. I the publicly available versions of unicall, and can't see how that could go wrong, even if you mix components from different versions. Now I can see what was my mistake. I was using the libraries: http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/ but there's no chan_unicall.c in there, so I took it from: http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/ and miss-patched the call events enum in unicall.h. I was using that mixture because I got some errors trying to compile unicall-0.0.3pre11. Now I solved the compile issue in unicall-0.0.3pre11 and Im using that, I can't still get it to work but I think it's a miss-configuration in some of the endpoints. I'll keep trying. Thnks.. -- Regards, Víctor Toofic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly MeetMe() question.
Hi, what does your modules directory contain? Can you find a file /usr/lib/asterisk/modules/app_meetme.so after make install? Knud Alex Balashov schrieb: I've got the ztdummy kernel module loaded and seem to have all the desired prerequisites in place, but Asterisk never seems to compile with MeetMe() application support enabled, nor does there appear to be a module I am failing to load that would contain this application. Is there something really obvious I am missing? Thanks, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Knud A. Müller ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Que Say Time
Hello All, I am almost done with my notifications system, but I am stuck with prompting the correct time. I went over the phpagi doc's, on how to say a given time using SAY TIME time escape digits. According to http://www.voip-info.org/wiki/view/say+time it say time is number of seconds elapsed since 00:00:00 on January 1, 1970, Coordinated Universal Time (UTC). Do I have to compute my time based on 00:00:00 on January 1, 1970 and then it will prompt correct time? What I am looking for is that, say the time on any given number of seconds. Anyone can help? Cheers, Nitesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly MeetMe() question.
On Mon, 30 Jul 2007, Knud Müller wrote: what does your modules directory contain? Can you find a file /usr/lib/asterisk/modules/app_meetme.so after make install? No. I know it needs to be compiled, but it is not being compiled no matter what I seem to do in the way of arguments to ./configure, installations of zaptel, etc. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 licenses installed - voicemail has noaudio...
I can't even hear the password prompts. Ahh... have you loaded the G729 sounds? Are you getting errors in the logs? John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 licenses installed - voicemail has noaudio...
I don't have any g729 sounds loaded.. they are just the gsm sounds... shouldn't asterisk do the conversion.. although at a license hit? On 7/30/07, John Faubion [EMAIL PROTECTED] wrote: I can't even hear the password prompts. Ahh... have you loaded the G729 sounds? Are you getting errors in the logs? John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom custom ring tones (slightly OT)
At 21:59 7/29/2007, Paul Hales wrote: I even got a Polycom here saying I'll be back which was funny for about an hour, then not funny at all. PaulH Kewwl! How do you get the .wav files into the Polycom? On Fri, 2007-07-27 at 12:36 +0800, James Andrewartha wrote: Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: PAGE_BEEP se.pat.ringer.13.name=Page Beep se.pat.ringer.13.inst.1.type=chord se.pat.ringer.13.inst.1.value=12 se.pat.ringer.13.inst.1.param=200 se.pat.ringer.13.inst.2.type=chord se.pat.ringer.13.inst.2.value=15 se.pat.ringer.13.inst.2.param=600 se.pat.ringer.13.inst.3.type=branch se.pat.ringer.13.inst.3.value=-2/ Alternatively, since you can use .wav files for ring tones, do people have any recommendations for where to find some distinctive rings? Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe through DeadAGI has changed to return -1 on Hangup
I have a support call AGI script that has been working
flawlessly for a couple of years now. It dumps the customer into a
MeetMe conference room, then dials a bunch of support engineers,
and connects anyone who accepts the call into the conference room.
The conference room is recorded. After the support call is over,
the recording is emailed to a list for quality control, etc.
It stopped working correctly on Jun 25th. Roughly on that date, I
upgraded to Asterisk 1.2.20 (I'm now on 1.2.23, and it hasn't
worked correctly on any version since 1.2.19).
What happens now is that when the MeetMe is exited normally (all
participants hang up), the AGI script simply stops executing. I
see no error messages on the CLI. I turned on agi debug, and I
see that MeetMe is returning res=-1. That is not supposed to
happen with DeadAGI (if I understand correctly), and it didn't
used to happen.
If I exit the MeetMe with the #, then I correctly get res=0,
and the script indeed continues to process correctly.
It seems to me that since 1.2.20, and continuing through today's
1.2.23, DeadAGI is behaving like AGI on a hangup of MeetMe.
Can anyone else confirm this, and if so, let me know what I can do
to revert it? This is the entire diff of the current app_meetme.c
with the one from 1.2.19, and it seems too innocuous to be the
culprit, but of course, it _is_ a hangup, so perhaps it's as
simple as reverting this one change?!?
[EMAIL PROTECTED] asterisk]# diff /usr/src/asterisk/apps/app_meetme.c
/usr/src/asterisk-1.2.19/apps/app_meetme.c
40c40
ASTERISK_FILE_VERSION(__FILE__, $Revision: 69894 $)
---
ASTERISK_FILE_VERSION(__FILE__, $Revision: 59360 $)
1299,1302d1298
/* If the channel wants to be hung up,
hang it up */
if (ast_check_hangup(chan))
break;
And here is the entire diff from res_agi.c:
[EMAIL PROTECTED] asterisk]# diff res/res_agi.c
/usr/src/asterisk-1.2.19/res/res_agi.c
44c44
ASTERISK_FILE_VERSION(__FILE__, $Revision: 71656 $)
---
ASTERISK_FILE_VERSION(__FILE__, $Revision: 54771 $)
572c572,579
ast_playstream(fs);
---
res = ast_playstream(fs);
if (res) {
fdprintf(agi-fd, 200 result=%d endpos=%ld\n,
res, sample_offset);
if (res = 0)
return RESULT_SHOWUSAGE;
else
return RESULT_FAILURE;
}
625c632,639
ast_playstream(fs);
---
res = ast_playstream(fs);
if (res) {
fdprintf(agi-fd, 200 result=%d endpos=%ld\n,
res, sample_offset);
if (res = 0)
return RESULT_SHOWUSAGE;
else
return RESULT_FAILURE;
}
1106c1120
return res = 0 ? RESULT_SUCCESS : RESULT_FAILURE;
---
return res;
Thanks in advance!
___
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Re: [asterisk-users] Calling to users in other asterisk servers
Hello, in your sip.conf do you have [yourprovider] username= fromuser= secret= host=another.server.com nat=yes . . . . and in your extensions.conf And the extensions.conf: ... exten = _X.,1,Dial,SIP/yourprovider ... Best Regards sip:[EMAIL PROTECTED] ) On 7/29/07, Ary Junior [EMAIL PROTECTED] wrote: Ok, my firewall port forward rules: TCP5004 - 5082192.168.254. 2 UDP5004 - 5082192.168.254. 2 TCP4569 192.168.254. 2UDP 4569192.168.254. 2UDP1 - 2192.168.254 . 2 And it dont works... Any configuration in special for make call the to users in another asterisk servers? Thanks very much!!! On 7/28/07, Carlos Rojas [EMAIL PROTECTED] wrote: Hello, Do you have porf forwardin for SIP protocol in your firewall? SIP: 5060 udp rtp 1 - 2 udp (default) and IAX2 4569 udp Best Regards Carlos Rojas On 7/28/07, Ary Junior [EMAIL PROTECTED] wrote: Hi, Im a asterisk newbie and I've configured an asterisk server here in my house... in my LAN two users can login and call to each other, but when I try to call an user in another asterisk server outside my LAN ( sip:[EMAIL PROTECTED] ) it dont work... if the person outside is conected on my server it works fine... My asterisk server is behind a firewall and portfowarding... it is possible? Thanks very much!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Huntgroup with asterisk feature
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 satish patel wrote: can you explain me how it will work caz i have not much idea about asterisk i am beginner so can u explain me how to use queue and how to forward my call to huntgroup http://www.orderlyq.com/asteriskqueues.html Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iD8DBQFGrgVvCFu3bIiwtTARAvSfAJ9YvE29ahDUC1oXz2xTXy6mORFjSgCeL4G+ q/GUEvk9raQV72ZMuuKaiP4= =pr5U -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream RTP keepalive packets causing Asteriskwarning
Hi Steve, The following packets match the offending warnings on the * console by time and number of occurrences and . The SRC and DST ports vary between calls (naturally) but the rest remains the same. Frame 3 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: Grandstr_0b:a5:3e (00:0b:82:0b:a5:3e), Dst: Dell_3e:05:ac (00:13:72:3e:05:ac) Internet Protocol, Src: 10.1.10.184 (10.1.10.184), Dst: 10.1.10.1 (10.1.10.1) User Datagram Protocol, Src Port: avt-profile-1 (5004), Dst Port: 13608 (13608) Source port: avt-profile-1 (5004) Destination port: 13608 (13608) Length: 16 Checksum: 0x8e5f [correct] Data (8 bytes) Data is 8 x null (00) regards, Drew Steve Langstaff wrote: Grab a network trace (with e.g. Wireshark) and look at the payload type and lengths of the RTP keepalive messages - if you post this information to the list I'm sure someone will comment on what's happening. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: 26 July 2007 19:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Grandstream RTP keepalive packets causing Asteriskwarning Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but it should not affect anything. In other words, Don't worry, be happy!. Any thoughts/experience on this? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions about SPA3102.
Hello, I got a SPA3102 and everything works fine except calling from voip to phone on fxo port. The phone ring but doesn't get any sound. I connected SPA at my asterisk server and i want to call from asterisk through SPA to fxo port where i have a regular phone. Thank you for support. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] String truncation problems on FreeBSD Sparc64
I've been investigating an issue on the Asterisk bugtracker recently: http://bugs.digium.com/view.php?id=10300 The reporter shows that there are places in the code where strings are truncated. You can read the bug report for full information. I suspect that the problem is specific to the reporter's system, but I want to be sure that this isn't standard behavior on certain hardware, OS, and compiler combinations. Are there any other users who have had this issue or one similar, and if so what OS, hardware, and compiler are you using? Thanks, Mark Michelson ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel channel reservation
Why would you want to do that? let Asterisk (using zap/g in app_dial) take care of which channel are used for outbound but assign all the channels to that g, reject any incoming calls if there are already 7 incoming active calls with a congestion PRI_CAUSE. Do the same for 20 outgoing active calls. That should solve your problem, that is if your problem is that you don't want to allow more than 20 outbound calls, and 7 inbound calls. There should be no need to reserve channels, in fact you could put in much better logic to accomplish what you want efficiently. For example although in general you want 20 outbound calls, but if currently 7 channels are still available (because there are no incoming active calls) you might want to allow up to x additional channels for outbound, until at least 3 incoming channels are active (or whatever the number). The same goes the other way around. Which you wont be able to accomplish by reserving channels. In fact by reserving channels, you might lose lots of incoming calls, while the outbound channels have never reached peak, which basically makes this whole thing a waste of money (on the unused channels). On 7/30/07, Jack [EMAIL PROTECTED] wrote: Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel = 1-15 channel = 17-21 ; outgoing group = 2 signalling=pri_cpe channel = 22-28 ; emergency group = 3 signalling=pri_cpe channel = 29-31 How can I avoid that incoming calls are going to the channels 22-31? Regards, Jack ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP Session Streaming
Hello, I am trying to transmit and receive sound over IP using the Java Media Framework(JMF) RTP. I was wondering if its possible to create an RTP Stream from my own computer and assign it to a URL. If anyone knows how I would do this, could they point me to some instructions or an example. So far, I have some sample code which compiles and creates the player, but it can't seem to realize the player, due to the URL that is given in the example. I'm guessing I need to create my own RTP stream to test this voice receiver. It would be nice if I could write the code for a voice transmitter and receive the RTP Stream from the transmitter using my voice receiver but I am not sure if this is possible on one PC. Would anyone be able to give me some advice on this matter. Thank you, Denis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound caller ID
Am Montag, den 30.07.2007, 05:24 -0700 schrieb Vieri: Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) 1) No 2) Depends. In some ISDN networks you can pay for an additional feature CLI not screened or similar, which means the number sent will not be corrected by the telco switching equipment if an invalid number is sent. AFAIK standard ISDN lines do not allow to send a number that is not connected to the line in question. 3) AFAIK same for ISDN PRIs. 4) Depends. Some allow sending any number, some always send your number and do not even allow to bar the number. sipgate.de for example allow to send any German regular number (or any number that looks like a valid number), but blocks special (0800, 0900, 112) numbers. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] software bloat - is this really useful to anyone?
http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
On Mon, 2007-07-30 at 14:29 -0700, Lee Howard wrote: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? This was added as a April Fools joke, and has since been removed. It's nice to know that even software engineers have a sense of humor from time to time. :-) On the serious side, one might see it as an example of how easy it is to add a new CLI command to Asterisk (complete with compiler flags, command auto-completion, and the whole nine yards). -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager - QueueAdd
Greetings all, When using QueueAdd via the dialplan app, we are able to define an agent name... however, I don't see how this can be done via the Asterisk Manager. Am I missing something, or is this just not possible? Regards, Jeff ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
Quoting Lee Howard [EMAIL PROTECTED]: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? I have to agree, there comes a time when someone has to say no to stuff that has no business being in production software. Remember when an o/s fit on a floppy with room to spare (and not just the bootstrap to load the real o/s) Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
Lee Howard wrote: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? Lee. It was done as a joke. It was committed only to trunk, and was only compiled if explicitly enabled. Mark seemed to get a kick out of it, so, yes, I guess you could say it was useful to at least one person. (Note, this code has since been removed, since clearly the date has long passed.) -- Jason Parker Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
Relax, its only in trunk. Zoa Lee Howard wrote: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
Jon Pounder wrote: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? I have to agree, there comes a time when someone has to say no to stuff that has no business being in production software. Well, you'll have to excuse him for trying to make a joke. :) Anyway, you'd be happy to know that this was never in a release, nor was it ever intended to be. It was only introduced in the development tree and has since been removed. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
On Mon, Jul 30, 2007 at 02:29:32PM -0700, Lee Howard wrote: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? Ghod... nobody has a sense of humour anymore. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
On Mon, 30 Jul 2007, Jay R. Ashworth wrote: On Mon, Jul 30, 2007 at 02:29:32PM -0700, Lee Howard wrote: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? Ghod... nobody has a sense of humour anymore. :-) Might just be making a slippery slope argument. Start with a birthday, and soon you have... COMMUNISM. This message has been brought to you by J. Edgar Hoover. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
Jay R. Ashworth wrote: Ghod... nobody has a sense of humour anymore. :-) I know. I better not list all of the other things we have done as a joke. Someone might have heart failure. ;) -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager - QueueAdd
try adding the line MemberName : name On 7/30/07, Jeff Iddings [EMAIL PROTECTED] wrote: Greetings all, When using QueueAdd via the dialplan app, we are able to define an agent name... however, I don't see how this can be done via the Asterisk Manager. Am I missing something, or is this just not possible? Regards, Jeff ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] software bloat - is this really useful to anyone?
On Mon, Jul 30, 2007 at 05:42:52PM -0400, Jon Pounder wrote: Quoting Lee Howard [EMAIL PROTECTED]: http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? I have to agree, there comes a time when someone has to say no to stuff that has no business being in production software. Remember when an o/s fit on a floppy with room to spare (and not just the bootstrap to load the real o/s) OT: Still fits on a floppy, basically: http://leaf.sourceforge.net/ If you look well enough you'll find Asterisk packages (e.g: in the contrib for Bering uClibc). But that asterisk.lrp doesn't fit in a floppy anymore. OTOH: /usr/sbin/asterisk depends on libopenh323 . Now this *is* bloat. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager - QueueAdd
After: Action: QueueAdd I presume? Thanks! 0xception wrote: try adding the line MemberName : name On 7/30/07, *Jeff Iddings* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Greetings all, When using QueueAdd via the dialplan app, we are able to define an agent name... however, I don't see how this can be done via the Asterisk Manager. Am I missing something, or is this just not possible? Regards, Jeff ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager - QueueAdd
That did the trick, thanks! Question, where did you find that documented? :) Jeff 0xception wrote: try adding the line MemberName : name On 7/30/07, *Jeff Iddings* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Greetings all, When using QueueAdd via the dialplan app, we are able to define an agent name... however, I don't see how this can be done via the Asterisk Manager. Am I missing something, or is this just not possible? Regards, Jeff ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with Speechphone
Has anyone set up Speechphone (Mandi) directly with Asterisk and not used an ATA? If so, could you share how you did it? TIA ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lightweight IAX balancer
Very low chances for that module if any. I haven't been using OpenSER much and I don't think I'll be using it soon - but who knows. Let's hope that implementation will be clean enough to turn it into a library easily if someone else wants to do it one day. So far another pack is available at previous link (http://www.gradwell.com/tmp/iax_proxy.tar.gz) with correct license banners (MIT). Plans for near future are: - stability - speed - POKE'ing and automatic dis/en-abling servers for balancing - for now you can reload list at runtime with SIGUSR1 without dropping calls - maybe some proper site for project... so I won't spam this maillist --- Stanisław Pitucha Gradwell Dot Com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager - QueueAdd
you know... i have no clue... i have it in my code somewhere. so i must of found it someplace. possibly in the phpagi-manager code. maybe some other random place. most asterisk info is scatter about, mixed up, and often out of date. so it's really really hard to tell some times. On 7/30/07, Jeff Iddings [EMAIL PROTECTED] wrote: That did the trick, thanks! Question, where did you find that documented? :) Jeff 0xception wrote: try adding the line MemberName : name On 7/30/07, *Jeff Iddings* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Greetings all, When using QueueAdd via the dialplan app, we are able to define an agent name... however, I don't see how this can be done via the Asterisk Manager. Am I missing something, or is this just not possible? Regards, Jeff ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager - QueueAdd
Aye. I'm not one to ask without doing a bit of research and I couldn't find that anywhere... even tried to figure it out by looking at the code. You're the best. Thanks again. Jeff 0xception wrote: you know... i have no clue... i have it in my code somewhere. so i must of found it someplace. possibly in the phpagi-manager code. maybe some other random place. most asterisk info is scatter about, mixed up, and often out of date. so it's really really hard to tell some times. On 7/30/07, *Jeff Iddings* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: That did the trick, thanks! Question, where did you find that documented? :) Jeff 0xception wrote: try adding the line MemberName : name On 7/30/07, *Jeff Iddings* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Greetings all, When using QueueAdd via the dialplan app, we are able to define an agent name... however, I don't see how this can be done via the Asterisk Manager. Am I missing something, or is this just not possible? Regards, Jeff ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
On Mon, 2007-07-30 at 16:09 +0100, Chris Blunt wrote:
If the PSTN is in use on SPA3102 I need a way to get the call to then
route out over IAX termination.
Usually, the best way to accomplish this is to send a call to your
Linksys ATA by using the Dial application from the dialplan, and then
looking at the result that gets set in the DIALSTATUS variable. For
example, you could try something like this:
exten = 123,1,Dial(SIP/linksys/5551212,30)
exten = 123,n,GotoIf($[${DIALSTATUS} = CONGESTION]?try-iax)
exten = 123,n,Busy(3)
exten = 123,n,Hangup()
exten = 123,n(try-iax),Dial(IAX2/my_iax_peer/5551212,30)
Obviously my example isn't that robust... it's simply meant to
illustrate the idea. (It depends on the SPA3102 returning a status code
that maps to CONGESTION if it's already in use... I don't have an
SPA3102, so I can't tell you how it actually performs.)
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] outbound caller ID
1. No 2. No 3. Only if your particular provider's switch allows it. Most will allow numbers to be set, but block the call if you try to set name. 4. Yes. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, July 30, 2007 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] outbound caller ID 1 No 2 I dont know. 3 Currently in the us the answer is yes On 7/30/07, Vieri [EMAIL PROTECTED] wrote: Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connections broken
Just so people on the list can search later: I found the solution: The smoothwall we have as our firewall / router needed to be reset. It went haywire and wasn't forwarding anything after about the 5th entry. I deleted everything out of the web interface for port forwarding, confirmed it went bye bye by ssh'ing into the box and actually looking at the files, restarted it, re-added the ports, and VIOIA! IAX works once again. What a pain in the asset. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Monday, July 30, 2007 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX connections broken On 7/30/07, Jared Smith wrote: Just for your information, IAX traffic is UDP, not TCP. I just thought I'd bring that up so that someone didn't mistakenly open up their firewall for TCP traffic instead of UDP traffic and wonder why IAX traffic wasn't making it through. Amen ! I had changed my router, the calls via my DID were working fine, but I just COULD NOT get either of my soft phones to connect. I looked at the contexts, nothing. The * console was not dead as ever. I check the port forwarding and Bingo ! only TCP was being forwarded. Aaaah ! -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe through DeadAGI has changed to return -1 on Hangup
Following up on my own post, and not quoting myself (tsk, tsk), I found a forum thread on Google that discussed a similar problem. They claimed it was a SIGHUP being sent to the script when the caller hung up, even though DeadAGI shouldn't get that type of signal. Anyway, it turns out that was my exact problem as well. I inserted a signal handler that ignores SIGHUP and my script now works the way it used to. This is for the next poor soul that trips on this problem... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel compiling broken: error: conflicting types for '__kernel_dev_t'
dear all, when i complied the latest Zaptel-1.2.19 to upgrade my asterisk system, it told me those errors: cc -c -fPIC -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE -o zonedata.lo zonedata.c In file included from zaptel.h:31, from tonezone.h:27, from zonedata.c:26: /usr/include/linux/types.h:18: error: conflicting types for '__kernel_dev_t' /usr/include/asm/posix_types.h:10: error: previous declaration of '__kernel_dev_t' was here /usr/include/linux/types.h:30: error: syntax error before timer_t /usr/include/linux/types.h:31: error: syntax error before clockid_t make: *** [zonedata.lo] Error 1 my environment: CenOS 4.2, gcc-3.4.4-2, kernel-2.6.12.2 my system is running asterisk-1.2.9.1+zaptel-1.2.6 can anyone give me some advise? thanks a lot. B/Rgz. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE212 or TE220
I am using TE212P with asterisk-1.2.18. It has echo DTMF in hardware to support. I use it on Dell Power Edge 85 no IRQ's ... Ya, just make sure that u get a good card I got the a broken card first time which ddnt work for echo cancellor then RMA'ed it with new one. -- Deepak fateme fatah [EMAIL PROTECTED] wrote: Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Description for each sound files
Hello Tzafrir, On 7/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: It's in /usr/share/doc/asterisk-sounds-main/sounds.txt.gz , as you should have expected (documentation for package foo normally resides at /usr/share/doc/foo/ and text files that are long enough are gzipped). I already checked that directory before I post this message to this mailing list because I was expecting that there will be a relevant information of what I've been looking for but it's not there. # ls -l /usr/share/doc/asterisk-sounds-main/ total 144 -rw-r--r-- 1 root root 16776 2007-06-20 01:16 changelog.Debian.gz -rw-r--r-- 1 root root 107576 2007-06-16 05:03 changelog.gz -rw-r--r-- 1 root root 12226 2007-06-20 01:16 copyright # dpkg -l | grep asterisk ii asterisk 1.4.5~dfsg-1 Open Source Private Branch Exchange (PBX) ii asterisk-config 1.4.5~dfsg-1 config files for asterisk ii asterisk-doc 1.4.5~dfsg-1 documentation for asterisk ii asterisk-sounds-main 1.4.5~dfsg-1 sound files for asterisk Should I build the Asterisk-1.4.9~dfsg-1 and related packages from the Debian Unstable for my Debian Etch box already? Please advice. Thank you. GNUbie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk or asterisknow
Hi: I want to have conference call service.You offer me use asterisk or asterisknow. Regards. - Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Royalty for On Hold Music ?
Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk or asterisknow
You can use both Asterisk or AsteriskNow to have meetme (conference room) On 7/30/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: I want to have conference call service.You offer me use asterisk or asterisknow. Regards. -- Be a better Globetrotter. Get better travel answers http://us.rd.yahoo.com/evt=48254/*http://answers.yahoo.com/dir/_ylc=X3oDMTI5MGx2aThyBF9TAzIxMTU1MDAzNTIEX3MDMzk2NTQ1MTAzBHNlYwNCQUJwaWxsYXJfTklfMzYwBHNsawNQcm9kdWN0X3F1ZXN0aW9uX3BhZ2U-?link=listsid=396545469from someone who knows. Yahoo! Answers - Check it out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
