[asterisk-users] disabling hardware echo can on tdm2400p
Hey list, Short version: I have a need to disable the hardware can on the tdm24xxp I have. I figure it's something in zconfig.h in the zaptel directory, but I'll be damned if I can figure it out. Long version: I have a tdm2403e card which is experiencing an odd problem; When several lines are in use, there is a bleeding of lines. My users call it the 'ghost'. Regardless, they can hear other people's conversation on different lines. I've been told this has to do with the hardware echo can I have on there, and that I should disable it if I continue having problems. So that's where I stand. Answers and opinions welcome. Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxy: one way audio
Hey all, So I just got an iaxy to play with a few days ago. Got the config files figured out and configured the device. I was able to make phone calls out on it just fine. However, when trying to call the device I get a one way audio problem ( which I would expect from sip, but not iaxy ). The user on the iaxy can hear but their audio isn't transmitted. I have double checked the iaxyprov config file, turning on heartbeat ( in case it's a firewall timeout problem ). I checked asterisk's iaxy.conf file, and all the ip information in there looks correct. I'm not sure how to procede to troubleshoot this problem. Any help is greatly appreciated. Sean iax260.conf: [EMAIL PROTECTED] trunk]# vi iax260.conf ; ; IAXY Provisioning description ; dhcp ;ip: 192.168.3.90 ;netmask: 255.255.255.0 ;gateway: 192.168.3.1 codec: ulaw ;codec: adpcm server: 192.168.1.7 ;altserver: 192.168.0.2 user: user pass: userpass register heartbeat ;debug ; ; Feature tuning (default is all enabled) ; ;disablecid ;disablecw ;disablecidcw ;disable3way iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 192.168.1.7; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [user] username=user type=friend secret=userpass record_out=Adhoc record_in=Adhoc qualify=no port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device user trunk=no ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxy: one way audio
Responding to my post for searching purposes; The fix is to manually specify disallow=all, allow=ulaw for each device. It does not seem to work if you only include that in the globals. Sean Sean Kennedy wrote: Hey all, So I just got an iaxy to play with a few days ago. Got the config files figured out and configured the device. I was able to make phone calls out on it just fine. However, when trying to call the device I get a one way audio problem ( which I would expect from sip, but not iaxy ). The user on the iaxy can hear but their audio isn't transmitted. I have double checked the iaxyprov config file, turning on heartbeat ( in case it's a firewall timeout problem ). I checked asterisk's iaxy.conf file, and all the ip information in there looks correct. I'm not sure how to procede to troubleshoot this problem. Any help is greatly appreciated. Sean iax260.conf: [EMAIL PROTECTED] trunk]# vi iax260.conf ; ; IAXY Provisioning description ; dhcp ;ip: 192.168.3.90 ;netmask: 255.255.255.0 ;gateway: 192.168.3.1 codec: ulaw ;codec: adpcm server: 192.168.1.7 ;altserver: 192.168.0.2 user: user pass: userpass register heartbeat ;debug ; ; Feature tuning (default is all enabled) ; ;disablecid ;disablecw ;disablecidcw ;disable3way iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 192.168.1.7; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [user] username=user type=friend secret=userpass record_out=Adhoc record_in=Adhoc qualify=no port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device user trunk=no ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxy will register, but doesn't detect POTS line
Wilson Pickett wrote: I'm thinking I have a faulty unit, but I would love to get some debug information out of it. Can anybody give me any pointers or suggestions on how to continue from here? I figure I'm calling digium come monday, but I would like to have it figured out by then. What phone are you trying to plug in to the iaxy? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Some cheap phone from target. It works on the other iaxy I bought; no issues, which leads me to believe that this is a iaxy problem, not phone problem ( or network, or power ). Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxy will register, but doesn't detect POTS line
So I got some iaxys in the other day. I got one of them working, the other is having issues. I am able to ping it, and upload a configuration file to it with a response. Afterwards, it even registers with the asterisk server. However, I am unable to get a dial tone, nor does the device seem to register the phone being picked up at all. I'm thinking I have a faulty unit, but I would love to get some debug information out of it. Can anybody give me any pointers or suggestions on how to continue from here? I figure I'm calling digium come monday, but I would like to have it figured out by then. Thanks in advance! Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxyprov downloading problems
Hi list, I just recently purchased some iaxy devices. Being new to this, I didn't have the iaxyprov tool, so I downloaded the instructions and attempted to follow them. Below is the problem I ran into. [EMAIL PROTECTED] src]# svn co http://svn.digium.com/svn/iaxyprov/trunk svn: 'trunk' is already a working copy for a different URL I'm not too familar with svn either, which is unfortunate. I used a web browser and browsed to that directory; Everything seems to be whre is should be. I could download each file manually, but I'm trying to avoid that kind of headache if possible. Can someone tell me what i'm doing wrong? Thanks Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxyprov downloading problems
Hadley Rich wrote: On Friday 22 September 2006 15:21, Sean Kennedy wrote: I just recently purchased some iaxy devices. Being new to this, I didn't have the iaxyprov tool, so I downloaded the instructions and attempted to follow them. Below is the problem I ran into. [EMAIL PROTECTED] src]# svn co http://svn.digium.com/svn/iaxyprov/trunk svn: 'trunk' is already a working copy for a different URL Looks like you already checked out Asterisk or something to the directory trunk in the current working directory. Try; mv trunk whatever then; svn co http://svn.digium.com/svn/iaxyprov/trunk iaxyprov to check out iaxyprov to the iaxyprov directory rather than trunk. hads Ha, I'm an idiot. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iaxyprov downloading problems
Hadley Rich wrote: On Friday 22 September 2006 15:21, Sean Kennedy wrote: I just recently purchased some iaxy devices. Being new to this, I didn't have the iaxyprov tool, so I downloaded the instructions and attempted to follow them. Below is the problem I ran into. [EMAIL PROTECTED] src]# svn co http://svn.digium.com/svn/iaxyprov/trunk svn: 'trunk' is already a working copy for a different URL Looks like you already checked out Asterisk or something to the directory trunk in the current working directory. Try; mv trunk whatever then; svn co http://svn.digium.com/svn/iaxyprov/trunk iaxyprov to check out iaxyprov to the iaxyprov directory rather than trunk. hads Ok, maybe i'm not retarded. I just tried that, and this is what I got ( in an empty directory mind you ): svn: REPORT request failed on '/svn/iaxyprov/!svn/vcc/default' svn: REPORT of '/svn/iaxyprov/!svn/vcc/default': 400 Bad Request (http://svn.digium.com) I'm not clear on what the error is, but I know both that directory exist and that I can ping that server. So i have no idea. Thanks for the help! Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxy configuration problems
Hi all, I followed the instructions found here: http://www.digium.com/en/docs/S101I/Iaxy_Installation_Guide.pdf when attempting to configure my iaxy. Sadly, it does not work. I upload the configuration file to the correct IP address, then unplug/plug the thing back in. I never see any registration attempts, and the orange/red light blinks every couple seconds. I can ping it, but that's about it. I'm not quite sure how to procede from here. Can anybody help? Thanks Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Centos cause Asterisk crash
chan, Run each script seperately to determine which one causes the crash. From there, check your logs to see any error messages. There should be something. My hunch is that prelink will cause the crash. chan (Alpha Trilogies Networks) wrote: Hi, Can some one who experience that does those file necessary for the CentOS and Asterisk installation /etc/cron.daily/00-makewhatis.cron /etc/cron.daily/slocate.cron /etc/cron.daily/prelink /etc/cron.daily/rpm /etc/cron.weekly/00-makewhatis.cron I experience that those file cause my Asterisk Server crash. Can I just disable them and run the Asterisk stable? Any reply will be appreciated. Thank you in advance. begin:vcard fn:Sean Kennedy n:Kennedy;Sean org:Rickey Wong DDS Inc adr;dom:A115;;2937 Veneman Ave;Modesto;CA;95356 email;internet:[EMAIL PROTECTED] title:Chief Information Officer tel;work:209-577-0777 x44 tel;fax:209-529-3209 tel;cell:209-485-2834 x-mozilla-html:TRUE url:http://www.qualitydentists.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues and the '*' key
[EMAIL PROTECTED] asterisk]# asterisk -V Asterisk SVN-branch-1.2-r8632M I was wondering if there was some documentation I was missing on the '*' key and queues. I have my features setup to use *x, where x is a #, but these features don't work for calls coming in from a queue. As soon as the '*' button is hit, the call is disconnected. I have a vague memory of reading about this somewhere, but searched @ the wiki AND through google aren't turning up anything useful. Sean begin:vcard fn:Sean Kennedy n:Kennedy;Sean org:Rickey Wong DDS Inc adr;dom:A115;;2937 Veneman Ave;Modesto;CA;95356 email;internet:[EMAIL PROTECTED] title:Chief Information Officer tel;work:209-577-0777 x44 tel;fax:209-529-3209 tel;cell:209-485-2834 x-mozilla-html:TRUE url:http://www.qualitydentists.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Menu in queue
Poul, I think what you are after is the context option in queues.conf. You can define a context with different options, then have your announcement detail the menu to your users. That's how I do it anyway. For much the same thing you use it for. Sean Poul Møller Hansen wrote: I'm wondering how I can let the caller choose to leave a voicemail message or continue to wait. Of course I can leave the queue and let the caller go back to the queue is he/she decides to stay waiting. But then they are new in queue again. How can I make such a menu where the caller keep their number in queue ? Thanks, Poul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reverse group in zapata.conf
Much thanks to you and the others who responded. Turns out I had been reading the part in the wiki that contained this a few times and completely missed it. My thanks Greg Scasny wrote: I think what your asking is pretty easy, just change the lowercase g in your extensions.conf file to an uppercase G. If you have a TRUNK type variable declared, this will be cake. If not you will need to change the little g, as in Zap/g1 to Zap/G1 everywhere you have it used. Hope that helped Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com [EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 - Toll Free 219-462-7257 - Fax. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Tuesday, March 07, 2006 8:04 PM To: Asterisk - Users Subject: [Asterisk-Users] Reverse group in zapata.conf Hey all, I have a situation where I have 8 lines from the phone company in a hunt group coming in to my asterisk box. These are the same lines I'm using for outgoing calls ( named g0 ). The problem arises when someone dials our number at the same time asterisk tries to put a call out on one of the zap channels in the g0 group. This has happened twice that I know of so far, once to myself. Asterisk opens the line before it's answered, and tries to dial. This has the effect of connecting the outside caller to the dialing party, which is the problem. My rather messy solution would be to have a reverse 'group' command in my zapata.conf file. So if I try dialing out on g1 ( my reverse group, 24-17 ), it starts at the top and works it's way down. Meanwhile, my external hunt group would still ring normally ( 17-24 ), thus minimizing the potential for conflict to a level that I'm comfortable with. Is this possible? If it isn't, I plan to reverse the order in which the lines are connected to my * box, having the same effect ( with no configuration changes. :) ). Anybody have any advice why I shouldn't do this either? Any other suggestions? Thanks Sean Kennedy begin:vcard fn:Sean Kennedy n:Kennedy;Sean org:Rickey, Wong DDS Inc adr;dom:Suite A115;;2937 Veneman Ave;Modesto;CA;95356 email;internet:[EMAIL PROTECTED] title:Chief Information Officer tel;work:(209) 338-0777 x44 tel;fax:(209) 529-3209 tel;cell:(209) 485-2834 x-mozilla-html:FALSE url:http://www.qualitydentists.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Snom 320, displaying text on the screen from *
Hey all, First of all, thank you for the help I've gotten on this list in the past. Very helpful, and I apprecaite it. Now, what I'd like to do is send a message to my snom 320s. I'd like to have the message display regardless of what the phone is doing. I have been trying SMS, or the sipsak method on the wiki but I have had no luck thus far. Does anybody have this working, and if so, can you give me a pointer? I imagine this will work the same for the 360s as well. Sean begin:vcard fn:Sean Kennedy n:Kennedy;Sean org:Rickey, Wong DDS Inc adr;dom:Suite A115;;2937 Veneman Ave;Modesto;CA;95356 email;internet:[EMAIL PROTECTED] title:Chief Information Officer tel;work:(209) 338-0777 x44 tel;fax:(209) 529-3209 tel;cell:(209) 485-2834 x-mozilla-html:FALSE url:http://www.qualitydentists.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Snom 320, displaying text on the screen from *
I have that set, but for some reason I get errors when I try sipsak, and nothing comes through to the phone: sipsak -M -B "test" -s sip:[EMAIL PROTECTED] timeout after 500ms timeout after 500ms... Some debugging info: [EMAIL PROTECTED] root]# sipsak -vvv -M -B "test" -s sip:[EMAIL PROTECTED] warning: ignoring -i option when in usrloc mode fqdnhostname: 192.168.1.1 our Via-Line: Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias New message with Via-Line: MESSAGE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 MESSAGE Content-Type: text/plain Max-Forwards: 70 User-Agent: sipsak 0.9.5 From: sip:[EMAIL PROTECTED]:34213;tag=7c8bd47f Content-Length: 4 test sending message ... request: MESSAGE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 MESSAGE Content-Type: text/plain Max-Forwards: 70 User-Agent: sipsak 0.9.5 From: sip:[EMAIL PROTECTED]:34213;tag=7c8bd47f Content-Length: 4 test send to: UDP:192.168.1.67:5060 : ignoring MESSAGE retransmission timeout after 500 ms So I am at a bit of a loss. Thanks for your help though, I apprecaite it. :) Colin Anderson wrote: Trick with Sipsak is you have to change the network port to 5060 or sipsak messages never hit the right port. In the web interface, Advaced Avanced Network Network identity (port): change that to 5060 and you should be good assuming you can figure out sipsak's nasty syntax. hth. begin:vcard fn:Sean Kennedy n:Kennedy;Sean org:Rickey, Wong DDS Inc adr;dom:Suite A115;;2937 Veneman Ave;Modesto;CA;95356 email;internet:[EMAIL PROTECTED] title:Chief Information Officer tel;work:(209) 338-0777 x44 tel;fax:(209) 529-3209 tel;cell:(209) 485-2834 x-mozilla-html:FALSE url:http://www.qualitydentists.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reverse group in zapata.conf
Hey all, I have a situation where I have 8 lines from the phone company in a hunt group coming in to my asterisk box. These are the same lines I'm using for outgoing calls ( named g0 ). The problem arises when someone dials our number at the same time asterisk tries to put a call out on one of the zap channels in the g0 group. This has happened twice that I know of so far, once to myself. Asterisk opens the line before it's answered, and tries to dial. This has the effect of connecting the outside caller to the dialing party, which is the problem. My rather messy solution would be to have a reverse 'group' command in my zapata.conf file. So if I try dialing out on g1 ( my reverse group, 24-17 ), it starts at the top and works it's way down. Meanwhile, my external hunt group would still ring normally ( 17-24 ), thus minimizing the potential for conflict to a level that I'm comfortable with. Is this possible? If it isn't, I plan to reverse the order in which the lines are connected to my * box, having the same effect ( with no configuration changes. :) ). Anybody have any advice why I shouldn't do this either? Any other suggestions? Thanks Sean Kennedy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 on a server behind a linux based stateful firewall
Hey Rich, As it turns out, this wasn't a firewall configuration problem at all. I had the firewall cofiguration nailed from the outset. This was just me not being used to how AMP works ( new to AMP ). Thanks though, I apprecaite the offer for help! Sean Rich Adamson wrote: I've got an * sitting behind a linux iptables firewall. I have an account with teliax. After entering the registration information accurately and restarting *, iax2 show registry shows a registered status on that connection. However, whenever I try to place a call, I get a "No one is available to answer at this time (1:0/0/0)" status returned from the dial attempt. When the same account is used on the firewall ( which I installed * to test this ), it works just fine. `tcpdump -i eth1 host voip-co2.teliax.com` on the trouble box shows packets coming and going to teliax without a problem. So I'm sort of at a dead end on how to figure this out. Everything looks like it's registered, and packets are flowing to the box and the box is able to get out ok. But for whatever reason, my calls aren't being completed. Can anybody give me another idea of where to look for the problem? Do an 'iax2 debug' while placing a call and analyze the output. Paste it into a posting here if needed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 on a server behind a linux based stateful firewall
Hi all, I've got an * sitting behind a linux iptables firewall. I have an account with teliax. After entering the registration information accurately and restarting *, iax2 show registry shows a registered status on that connection. However, whenever I try to place a call, I get a No one is available to answer at this time (1:0/0/0) status returned from the dial attempt. When the same account is used on the firewall ( which I installed * to test this ), it works just fine. `tcpdump -i eth1 host voip-co2.teliax.com` on the trouble box shows packets coming and going to teliax without a problem. So I'm sort of at a dead end on how to figure this out. Everything looks like it's registered, and packets are flowing to the box and the box is able to get out ok. But for whatever reason, my calls aren't being completed. Can anybody give me another idea of where to look for the problem? Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone Recommendation
Kris, I highly recommend the snom 320. Very easy to configure, and very easy to setup line appearances. As has already been mentioned, the idea of lines is a bit dated. For more information, read this: http://forums.digium.com/viewtopic.php?t=891. Sean Duracom ISP Lists wrote: We are going to replace our existing PBX system with an Asterisks box. I have 7 phone lines that come in and I need to get a phone that would support that many lines at minimum. Do you guys recommend any phones that you have used that work well. Kris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] traffic shaping
Jose, I don't know what everyone else uses, but I use wondershaper. It's a bit rough, but it does the job well for what I need ( prioritize voip traffic over everything else ). Google should bring it up for you. Sean Jose Limeres wrote: Hi all, Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth reservation policy in *. What about using * with 2 network cards betwen the LAN and ADSL router and giving preference to VoIP traffic over web surfing? Thanks, jose ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions and regular expressions ( probably an easy question )
Hi all, I'm having a hard time finding information related to the regular expressions that can be used in a dialplan, specifically as an extension. For example, I have an 800 number which I'd like to jump directly to if my users dial it, instead of going over my pstn termination. Currently, it looks like this: exten = 8661234567,1,Goto(800-in) However, I'd like 1866123456 to match as well. I can't find in the wiki or sample configs how to say match this 0 or 1 times. Can anybody provide a link that would go over this? Again, I've been digging through the wiki, but I seem to be missing it. Thanks Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions and regular expressions ( probably an easy question )
Hi Dan, Thanks for the info, but what I'm after is the ability to match a digit/character 0 or 1 times at the beginning of the string. If I'm reading your example right, it'll match anything starting with 866, which doesn't work for me. I am trying to match: 18661234567 and 8661234567 Sean ps: The pdf doesn't have a good explaination of this either, although it occurs to me that this might not be possible with * if I'm having such a hard time finding it. Daniel Wright wrote: Sean Kennedy wrote: Hi all, I'm having a hard time finding information related to the regular expressions that can be used in a dialplan, specifically as an extension. For example, I have an 800 number which I'd like to jump directly to if my users dial it, instead of going over my pstn termination. Currently, it looks like this: exten = 8661234567,1,Goto(800-in) However, I'd like 1866123456 to match as well. I can't find in the wiki or sample configs how to say match this 0 or 1 times. Can anybody provide a link that would go over this? Again, I've been digging through the wiki, but I seem to be missing it. Thanks Sean You could do it like this: exten = _866.,1,GoTo(800-in) The period means match one or more characters. You can find reference to expressions and how they work in this pdf book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions and regular expressions ( probablyan easy question )
Steve, Yeah, that's what I've been doing, but I was hoping to make it a little clearer in the dial plan. Ah well, you win some and lose some. Thanks! Sean Steve Totaro wrote: You just need separate extensions. Hi Dan, Thanks for the info, but what I'm after is the ability to match a digit/character 0 or 1 times at the beginning of the string. If I'm reading your example right, it'll match anything starting with 866, which doesn't work for me. I am trying to match: 18661234567 and 8661234567 Sean ps: The pdf doesn't have a good explaination of this either, although it occurs to me that this might not be possible with * if I'm having such a hard time finding it. Daniel Wright wrote: Sean Kennedy wrote: Hi all, I'm having a hard time finding information related to the regular expressions that can be used in a dialplan, specifically as an extension. For example, I have an 800 number which I'd like to jump directly to if my users dial it, instead of going over my pstn termination. Currently, it looks like this: exten = 8661234567,1,Goto(800-in) However, I'd like 1866123456 to match as well. I can't find in the wiki or sample configs how to say "match this 0 or 1 times". Can anybody provide a link that would go over this? Again, I've been digging through the wiki, but I seem to be missing it. Thanks Sean You could do it like this: exten = _866.,1,GoTo(800-in) The period means match one or more characters. You can find reference to expressions and how they work in this pdf book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions and regular expressions ( probably an easy question )
Rich,
It's kind of tough to truly understand what you are trying to
accomplish
Ack, sorry! It's hard to post to the list on a saturday when my
2year old is wanting to play with the keyboard as well. Best I can do
is half a mind, most of the time that's enough.
Not always, however. :)
(or ask for). Apparently you've got something more in mind that
words are making it through the list. Reading between the lines, it
would appear from the 800-in that calls are coming in from some
external source, and you trying to do something with them. Can you be a
little more explicit
I have an 800 number from teliax. When my "local" users dial it, they
will dial 1866... instead of the 866 I have in my dial plan. I do not
want the call to use one of my external sources to terminate the call (
in essence, dialing out via voicepulse, and recieving the call via
teliax ). I know I can do two seperate exten patterns, but I was
hoping for a single pattern. To that end, I was wondering if there was
a way of saying "Match this 0 or 1 times", something I'm used to in
perl and the like.
If there isn't, there isn't. Won't kill me to add the second exten
match.
Sean
Rich Adamson wrote:
Or, just
do...
exten = 18661234567,1,Goto(800-in)
exten = 8661234567,1,Goto(800-in)
It's kind of tough to truly understand what you are trying to
accomplish
(or ask for). Apparently you've got something more in mind that words
are making it through the list. Reading between the lines, it would
appear from the 800-in that calls are coming in from some external
source, and you trying to do something with them. Can you be a little
more explicit.
Hi Dan,
Thanks for the info, but what I'm after is the ability to match a
digit/character 0 or 1 times at the beginning of the string. If I'm
reading your example right, it'll match anything starting with 866,
which doesn't work for me. I am trying to match:
18661234567 and 8661234567
Sean
ps: The pdf doesn't have a good explaination of this either, although
it occurs to me that this might not be possible with * if I'm having
such a hard time finding it.
Daniel Wright wrote:
Sean Kennedy wrote:
Hi all,
I'm having a hard time finding information related to the regular
expressions that can be used in a dialplan, specifically as an
extension. For example, I have an 800 number which I'd like to jump
directly to if my users dial it, instead of going over my pstn
termination. Currently, it looks like this:
exten = 8661234567,1,Goto(800-in)
However, I'd like 1866123456 to match as well. I can't find in the
wiki or sample configs how to say "match this 0 or 1 times".
Can anybody provide a link that would go over this? Again, I've been
digging through the wiki, but I seem to be missing it.
Thanks
Sean
You could do it like this:
exten = _866.,1,GoTo(800-in)
The period means match one or more characters.
You can find reference to expressions and how they work in this pdf
book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
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[Asterisk-Users] snom 320 'retrieve' button
Hi all, Got the snom 320s, and I love them. Only issue I'm having with setup is getting the retrieve button working. I have specified in my sip.conf: [EMAIL PROTECTED] Which is my Checking voicemail extension. However, when I hit the 'Retrieve' button, it seems sporadic what it dials. Some times it will try to dial [EMAIL PROTECTED] but get an error that I can't trace. Most of the time, it will dial the local extension. Very odd. For those of you who have this setup, what am I missing? How did you get this working? Thanks! Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 320s and the hint priority
Hi everyone, Does anyone have this working? I'm looking at these phones for my receptionist phone, with the requirement that the two bars of buttons and lights on the side show line presence for programmable extensions ( ie: line 1 show the presense of my 101 user, line 2 = 102 user, ect.. ). I don't want to buy them only to find they can't do this, so I was hoping someone on the list had these suckers up and running. Thanks Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 320s and the hint priority
Hey, you are my new best friend. I have never had a phone to use with the hint priority, would you mind giving me a sample of your configuration so I can figure it out? Much apprecaited! Sean Michiel van Baak wrote: On 10:25, Wed 30 Nov 05, Sean Kennedy wrote: Hi everyone, Does anyone have this working? I'm looking at these phones for my receptionist phone, with the requirement that the two bars of buttons and lights on the side show line presence for programmable extensions ( ie: line 1 show the presense of my 101 user, line 2 = 102 user, ect.. ). I don't want to buy them only to find they can't do this, so I was hoping someone on the list had these suckers up and running. Works great here :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Blind transfer question
Hi all, I'm trying to change the keys associated with the blind transfer function. I've been mucking around in features.conf, but nothing I do seems to make any difference ( and I've tried to intentionally break it ). I have restarted the * server between each modification. Is this a known thing? Can anybody give me an idea of how to change the Blind Transfer key sequence to something else? Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clearwire and Asterisk
Justin, I can tell you that I haven't been able to get a clearwire rep out to my location to demonstrate their lines to us. I keep calling, telling them what i want to do. And they tell me that clearwire is a great service, and that one of they will relay my questions to a sales rep who will get back in touch with me soon. And I never hear from them again. In my mind, that doesn't seem like a 'good thing'. YMMV Sean Justin Newman wrote: Has anyone had problems using Clearwire, VOIP, and/or Asterisk? Just curious... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse down?
They were down and now back up. Sean John L. Magee wrote: no DNS resolution to begin with. Anyone heard anything about this? jlm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Console ALSA Sound
Would see everyone else has already answered your question, so let me give you some background on it. When * loads you can see ( if you do verbose that is ) all the modules it's loading. Stock * loads more than I use, so I went through and wrote down all the modules I wasn't going to use ( or thought I wasn't going to use in one or two cases ) and then used those with the noload statement in the modules.conf file. Same principle applies here. Sean Conrad Beckert wrote: Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference FM is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Console ALSA Sound
Conrad, Heh, try asking about line appearances and the hint priority. People clam right up. Or ask about receptionist phones that show all your line statuses. You can practically hear the crickets. :) Sean Conrad Beckert wrote: To all who have answered my question: What a great mailing list - not even 1 hour wait and yet such a lot of qualified answers! Thank you very much Conrad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATTN: Keith
Legitimate or not, if you are asking me for help off list ( as a number of you have done ) and you never get a reply from me, that's likely why: You mail server is blocking mine based on a blacklist like ORBS. I don't waste any time trying to get around this, so I'm sorry if you folks never get the help you ask for, but it's something you are doing to yourselves. Sean list wrote: according to RFC's your required to have reverse lookups on ur mail server, so blocking based on this is perfectly legitimate. -jon - Original Message - From: Sean Kennedy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, June 09, 2005 2:28 PM Subject: Re: [Asterisk-Users] ATTN: Keith Matt wrote: I apologize for sending this to the list. Keith from Hazleton... your mail server is rejecting mail I'm sending you from my mail servers, as well as from gmail... you may really want to consider using a different blacklist.. the on you are using now is going to block almost everything and everyone. Honestly, when I've tried to reply to people who have contacted me off list, and I get a bounce because of a too restrictive black list, I just let it drop. ORBS is blocking my mail server for being on a dynamic address, for example. And given that I can't fulfill their requirements to get myself removed ( basically, I'd have to get my reverse to look proper or something ), I will always be on their blacklist. Just something to keep in mind, all of you using ORBS. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATTN: Keith
Matt wrote: I apologize for sending this to the list. Keith from Hazleton... your mail server is rejecting mail I'm sending you from my mail servers, as well as from gmail... you may really want to consider using a different blacklist.. the on you are using now is going to block almost everything and everyone. Honestly, when I've tried to reply to people who have contacted me off list, and I get a bounce because of a too restrictive black list, I just let it drop. ORBS is blocking my mail server for being on a dynamic address, for example. And given that I can't fulfill their requirements to get myself removed ( basically, I'd have to get my reverse to look proper or something ), I will always be on their blacklist. Just something to keep in mind, all of you using ORBS. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice - Customer feedback
Hi all, Can any broadvoice customers give me their opinions on the service recently? It's actually been pretty quiet on the list lately regarding them, so it seems to me that they're either getting things straightened out or everyone has dropped the service. Thank you in advance Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] `hint` priority and Polycom 500
Hi all, I'm trying to see if I can get the hint priority working with my polycom 500. So far I have 2 /reg entries with the same sip registration, one is labeled as private, the other as shared. I have set the hint priority before anything else in my dialplan for my extensions. As it stands, I have two registrations on the phone, one has a half greyed out phone icon, the other is a full icon. However, when I place a call to that phone, the shared line display doesn't change. I'd apprecaite any input people may have on this. Thanks Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is SKYPE a threat or should we do something (together)
Ronald Wiplinger wrote: Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald I think we already do this; DUNDI. There's even a service that allows unused lines to be used by others ( Hopefully someone can fill in the blanks here. I don't remember the name ). However, I don't think Skype and * have the same goals. They are, essentially, two different solutions to two different problems. At least, that's how I see it. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving user progress in an voice menu system
Hi Josiah
Thanks for the info. What I decided to do instead was to modify my own
macro so I could pass the ring type to it. It may have helped me had I
remembered that the default config comes with a dial macro, but then
probably not, as I rewrite things all the time.
I like to reinvent the wheel.
Anyway, I didn't think much about it at the time, because the problem
got solved, but I'll post my macro here in case it's useful to anybody
else. Keep in mind it's fairly limited in scope, and it depends on
outside variables that are common throughout my extensions.conf file. I
can't see it being terribly useful for anybody here, but what do I know?
[macro-ext]
;; ${ARG1} is the sip channel to dial, ${ARG2} is the dial type.
;;Most times, that's simply a ring. For the menu system, I have it play
music on hold while it tries an extension.
exten = s,1,Dial(${IN_CHAN}/${ARG1}|${IN_TO}|${IN_OPT}${ARG2})
exten = s,2,VoiceMail(su${ARG1})
exten = s,3,Hangup
exten = s,102,VoiceMail(sb${ARG1})
exten = s,103,Hangup
Josiah Bryan wrote:
On Thursday 12 May 2005 3:43 pm, Sean Kennedy wrote:
Hi all,
I have a voice menu system ( Outlined below ), and I'd like to give the
user some feedback when they dial an extension ( ringing, music,
SOMETHING ). As it stands, when a user enters an extension from the
menu system, they hear silence while the line rings. I even tried
including the Ringing application before calling my macro to dial the
phones, with no luck.
Any help is apprecaited.
Odd - my receptionist was having a similar problem. I used the stdexten macro
that came with the demo files - when ever someone dialed directly (inside) or
directly thru the IVR (no receptionist pickup) - the ringback was fine. But
when the receptionist picked up and transfered - no ringback. All three
methods of dialing went thru the stdexten macro - very puzzling. The solution
I finally came up with was to add the 'm' option to the 'Dial' command.
Code speaks louder than words, so here you go..its obviously modified a bit -
but all should be self explanitory. The SIP/op channel is our receptionist
phone. The macro only adds the MOH option if the call is from the
receptionist phone, otherwise it leaves all options at default.
Anybody else have any other solutions or need debug outputs to figure this
out?
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[Asterisk-Users] Voicepulse problems?
Hi all, Is anybody else experiencing problems with voicepulse? Today and over the weekend? I've had problems with both gateways, but one usually works when the other doesn't. I'm trying to eliminate my network from the problem. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Giving user progress in an voice menu system
Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is apprecaited. Sean [800-in] exten = s,1,Answer exten = s,2,Background(billing-welcome) exten = s,3,ResponseTimeout(5) exten = s,4,Background(billing-menu) exten = t,1,Goto(s,3) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,2) exten = 101,1,Ringing exten = 101,2,Wait(1) exten = 101,3,Macro(ext,101) exten = 113,1,Ringing exten = 113,2,Wait(1) exten = 113,3,Macro(ext,113) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving user progress in an voice menu system
Hi all, Thanks Christian for informing Dial() options which will let me do exactly what I am after. You'd think I'd remember things like that. *shakes head* Thanks again Sean Sean Kennedy wrote: Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is apprecaited. Sean [800-in] exten = s,1,Answer exten = s,2,Background(billing-welcome) exten = s,3,ResponseTimeout(5) exten = s,4,Background(billing-menu) exten = t,1,Goto(s,3) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,2) exten = 101,1,Ringing exten = 101,2,Wait(1) exten = 101,3,Macro(ext,101) exten = 113,1,Ringing exten = 113,2,Wait(1) exten = 113,3,Macro(ext,113) ___ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Polycom SP 500 and Cisco PIX
Hi Eduardo, You can reboot the phone by holding down +/-/Hold/Messages buttons. I'm not sure I know the problem with dhcp, it's definitely odd. Can you tell me what firmware you are running? Sean Eduardo Jimenez wrote: Hi everyone, Im very new to all this, so please forgive me if I have the terminology mixed-up. We are preparing to install an Asterisk IP PBX over the weekend and I have an issue with the Polycom SP 500 phones we are trying to use. My problem is regarding DHCP. Our DHCP server is our Cisco PIX 501 firewall. Ive specified option 66 and the phones connect to the FTP server when they boot. BUT, after theyve loaded sip.ln, a message comes up stating that the phone could not get an IP address from the DHCP server. Theres an about button right there, and it says it has an address, which is also ping-able. The phone then waits a little and reboots again. Switching to a fixed IP makes the phone boot all the way. I tried putting together a quick test network with a DHCP server on a BSD machine (using bootpd) and at least I got to the welcome screen (then the phone hanged and the reboot keystroke did not work).but I think that last thing is just an issue with everything else not being setup on the phone itself. Any ideas? Thanks, Eduardo Jimenez Software Engineer Job Performance Systems, Inc Ph: 703-683-5805 x 210 www.jps-usa.com http://www.jps-usa.com/ Also, I unpacked a phone, played with it for a minute and now that phone does not seem to boot. Tried the 4 6 8 * combination for about 10 secs and did not do anything.is there a harder reset than that? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse down?
Trevor Harrison wrote: Anyone else using Voicepulse? This morning I noticed that they seem to be doa... no dns resolution, no ping, etc. -Trevor Nope, working fine here ( Modesto California ). Try reversing which gateway you are using first. I did that a while ago and things seem to work fine now. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Live Voip
Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and calls on hold
Jeroen Moetwil wrote: Hello - I recently offloaded some of the SIP traffic on to a seperate Asterisk box and interconnected our main Asterisk system with the new system via IAX. The SIP clients are running 7960's. When a call is put on hold, often times when the call is pulled off hold, there seems to be no RTP in at least one direction. There seems to only be voice in one direction. Basically the call comes in via a ZAP channel over a PRI into our main system, is fed over IAX to our second system and then is connected to the SIP channel (client). I've tried both enabling and disabling IAX trunking and jitterbuffers. I've also added a zap card and enabled it to allow for a timing source. The new system is running the latest CVS of Asterisk and libraries as of yesterday, while the other one is running a CVS version as of Jun of last year. I'm using RSA for auth between the servers (IAX). Any help would be appreciated. Thanks. Jeroen Jeroen, I am by no means a guru, so take what I saw with a healthy sized grain of salt. You say you have two * boxes, connected with IAX. Are they both on the same subnet? My thoughts are this: The first box is trying to directly establish a route to the sip device, bypassing the SIP concentrator ( the second * box ). Again, I am probably wrong, but that's the only thing I can think of that would cause problems. The only times I've had problems with SIP and one way audio was across a vpn/nat system, so that might be something you have to take into account as well. In fact, now that I am thinking about it, if you haven't already I'd check the sip.conf file and make sure the bind address is correct. Hope some of that helped a little bit. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HINT
Anton Krall wrote: Guys, what does hint do in a dialplan and how do you use it? I have been trying to figure this out for a while now, even posted a question on the list, to which no one replied. Any details would be apprecaited if you find this one out. I want to use it, but I don't know how. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Quality
[EMAIL PROTECTED] wrote: Hello, I have setup two * servers and they are communicating using IAX. I'm passing calls from SRV A (internet connection T1) to SRV B (internet connection: 512). For some reasons I have an issue with the quality. The voice is a bit scratchy. I have tried iLBC and SPEEX, but it didn't make any difference. Now, assuming that I have an issue with Bandwidth, what would be the best way to configure my iax.conf. (A bit confused about jitterbuffer and tos) Here is my iax.conf @ location A: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay and iax.conf @ location B: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay [guest] type=user context=default callerid=Guest IAX User disallow=all allow=ilbc Thanks guys Have you tried ulaw yet? With 512 and a t1, you have more than enough bandwidth for a few streams with that codec. One wouldn't be a problem. Also, check to make sure the entire path is a single codec, I have run into an issue before ( when I first started playing with * as a matter of fact ), where I was going from gsm, to ulaw to alaw ( long story ) back to gsm. Voice quality sucked, obviously, because I was doing all sorts of conversions. Keep yourself to a single codec, preferrably ulaw/alaw, and you should be fine. Also check for iax timing, that could cause issues as well. TOS is a quality of service bit on the packets in the stream, they don't do anything by themselves. Instead, any switches/routers than understand it will push those packets to their appropriate position in the queue based on the TOS value. I'm not entirely clear on what jitter is either, but it's never been important enough for me to go digging. Anybody have any insight here? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please find me a IAX provider
Kumara Jayaweera wrote: Hi all, I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I don't need incoming calls (no need DIDs). Could someone tell me who is the best IAX service provider for me? I want unlimited monthly basis or yearly basis service. my DSL is 128kbps. Thank You Kumara Well, I dig voicepulse, but I don't know what kind of latency you'd be running into. Check out: connect.voicepulse.com. I don't know what the proper etiquette is regarding their iax2 server addresses, so if you want them to ping them, I'd ask them. I'm sure they'd give them to you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
Adam Goryachev wrote: On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote: 2) There isn't anything like what you want. I know, I want the same thing. There is no phone out there that will do this with any protocol that asterisk uses. This is the one major failing of asterisk ( and voip in general. I smell an oportunity for a phone manufacture ), and what keeps it out of a lot of places. It's alright, you can come out from under your rock now The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) phones can all do what he wants. ie, have multiple lines with blinking red lights when a call arrives on that line. The polycom ip600 and cisco 7960 both have 6 lines available. Regards, Adam Ok, this is the first I've heard about it. Will the lights show call status? As in, if the call is put on hold on one of those other extensions, it will flash? Or go green ( or another color ) when a call is connected on another extension? Basically a mimic of the partner ACS systems? To my knowledge, there is no such thing. Am I wrong? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hint priority...how does it work?
So, from what limited info I was able to find in the wiki, the hint priority should let another phone know when an extension is in use. So if we had this... exten = 1000,hint,SIP/101 when ext 1000 is dialed, SIP/101 can see it..somehow. how does this work, exactly? Can anybody provide a bit of background and depth to this? Thank you Sena ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
Jason Brown wrote: I have a problem. The average person is too freaking stupid to use a VOIP phone. My experience has so far been that if it doesn't have 20 buttons with little red LED's on it, the user cannot comprehend call parking, attended transfer, blind transfer, DND, and navigating through a voicemail menu. I need a good receptionist phone that works with Asterisk. It basically needs to act like an avaya partner phone, I don't need 20 buttons with little red LED's...what I do need is for the phone to register multiple extensions to my asterisk server and act like each SIP extension is a line, so if the idiot receptionist has a call ringing in on line 1, she can pick it up, look at the buttons, see a call ringing in on line 2 (and the phone ringer rings), put call 1 on hold without hanging the caller up, and hit the little I am an idiot and need a line 2 button to pick up line 2, so on and so forth. I love VOIP systems and all the functionality they bring and features I get. Unfortunately, the average person in this country anymore is apparently completely stupid and cannot understand how to juggle calls without hanging up on people. /rant So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy. 1) I suggest you learn to live and like those idiots. I also suggest you tone down that attitude and adjust it. Those idiots contribute to YOUR pay. 2) There isn't anything like what you want. I know, I want the same thing. There is no phone out there that will do this with any protocol that asterisk uses. This is the one major failing of asterisk ( and voip in general. I smell an oportunity for a phone manufacture ), and what keeps it out of a lot of places. I can see this being implemented with a phone that speaks to *'s manager interface. Who wants to talk to polycom or cisco about it? :) Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 800 number provider suggestions
Hi all, Can anybody recommend a good 1-800 number provider? Thanks Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Eric Wieling aka ManxPower wrote: Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; Subquestion to this ( although I much prefer setting the offset in the ipmid.cfg file myself ): How do you specify a negative offset when you are using the dhcp server that comes with windows server? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fail over solutions
Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Phones - No Audio In Either Direction
Paul Tyreman wrote: Hi, After months of testing Asterisk, I am finally ready to roll it out, replacing my previous VOIP server (brekeke's ondo SIP Server), which was very restrictive. However, I am experiencing some problems with phones which are on a different network to the server (connecting via the internet). I have managed to get the phone to register with the Asterisk server, and I can make a call and hear it ringing, but once connected no audio can be heard in either direction. I have opened the following ports: 5004, 5060, 5061 and 1 - 10010 on my router, but am still having no joy. When I used ondo, I had to add my WAN IP address to the configuration files, so I was wondering if I have to do that in some .conf file in Asterisk ? Hope someone can help ? Thanks, Paul. I had this problem on a vpn ( highly recommended, given how easy it is to implement openvpn now ). I changed the IP address in the SIP file to my server ( 192.168.1.1, remember, on a vpn ), everything just worked. Good luck Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Config - SIP 1.4.1
Wiley Siler wrote: Does anyone have an example of a working config for SIP 1.4.1? I made the transition and the phones seem to hate the new config file from the example. W Here are my polycom config files relating to sip ( names have been changed to protect the innocent ) Sean ?xml version=1.0 encoding=UTF-8 standalone=yes? !-- Example Per-phone Configuration File -- !-- $Revision: 1.59 $ $Date: 2004/05/22 00:50:41 $ -- phone1 reg reg.1.displayName=username reg.1.address=username reg.1.label=username reg.1.type=private reg.1.auth.userId=username reg.1.auth.password=password /reg msg msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=voicemailMain/ /msg /phone1 ?xml version=1.0 standalone=yes? !-- SIP Application Configuration File -- !-- $Revision: 1.57.2.1 $ $Date: 2004/07/27 00:23:34 $ -- sip voIpProt local voIpProt.local.port=5060/ server voIpProt.server.1.address=192.168.1.1 voIpProt.server.1.port=5060 voIpProt.server.1.transport=UDPonly voIpProt.server.1.expires=3600 voIpProt.server.1.register=1 voIpProt.server.1.retryTimeOut=0 voIpProt.server.1.retryMaxCount=0 voIpProt.server.1.expires.lineSeize=30/ SIP voIpProt.SIP.useRFC2543hold=1 voIpProt.SIP.lcs=0 voIpProt.SIP.sendCompactHdrs=0 voIpProt.SIP.WM50=0 voIpProt.SIP.keepalive.sessionTimers=0 voIpProt.SIP.requestURI.E164.addGlobalPrefix= outboundProxy voIpProt.SIP.outboundProxy.address= voIpProt.SIP.outboundProxy.port=5060/ alertInfo voIpProt.SIP.alertInfo.1.value=AA voIpProt.SIP.alertInfo.1.class=3/ alertInfo voIpProt.SIP.alertInfo.2.value=RA voIpProt.SIP.alertInfo.2.class=4/ requestValidation voIpProt.SIP.requestValidation.1.request= voIpProt.SIP.requestValidation.1.method= voIpProt.SIP.requestValidation.1.request.1.event= digest voIpProt.SIP.requestValidation.digest.realm=192.168.1.1/ /requestValidation specialEvent voIpProt.SIP.specialEvent.lineSeize.nonStandard=1 voIpProt.SIP.specialEvent.checkSync.alwaysReboot=0/ conference voIpProt.SIP.conference.address=/ /SIP /voIpProt dialplan dialplan.impossibleMatchHandling=2 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=911|[2-8]xx|1xx dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address= dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan logging level change log.level.change.sip=4 log.level.change.sip.obs=5/ /level /logging /sip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
JD Austin wrote: I tried calling Broadvoice support.. on hold for 1/2 hour then it hung up on me with a reorder (fast busy), so I tried again. Just got through to a rep- they said it's a 'carrier issue' that their 'partner carrier' was having issues and that it would be up soon. Makes me wonder if I should be signing up with their 'partner carrier' instead. JD Actually, with all the threads I've seen on the mailing list, I'm weary of anything having to do with broadvoice. Personally. Maybe it's just that they have such a large user base on linux. Who knows. Voicepulse gets my business tho. :) Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom ip500 (Not-Registered)
Dan Morin wrote: I just got a few Polycom IP500s and Ive been following the info in the wiki trying to configure them. From what I can tell, they seem to be setup correctly (wellthey dont work so obviously not) however, when they try to register with Asterisk, the following error shows up in the Logs: Apr 25 15:22:05 NOTICE[1718]: Registration from '' failed for '192.168.0.222' Apr 25 15:22:20 DEBUG[1718]: Auto destroying call '[EMAIL PROTECTED]' Where 192.168.0.222 is the IP of the phone. The two single quotes seem to indicate that no credentials are being passed to * (?). If anyone has any experience with these, please let me know. I can post the configs if that would help. Thanks in advance. Dan Please do. Specifically, sip.conf ( or whatever your sip configuration file for the phones is ), the individual phone settings and your sip.conf file from asterisk ( relevant parts only ). Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BYOD provider other than broadvoice
Michael Lyszczek wrote: Are there any BYOD providers out that that people have had positive experiences with? I have broadvoice and they suck lately. Anyony have anyone with a good amount of peers and not a lot of downtime? I like voicepulse. They raised their rates recently, but they are still reasonable and I haven't had any problems with them since I started using them back in November. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring two extensions at the same time
G.Marshall wrote: Hello, I can not find anything on this, so it may not be possible. I would like to dial one number which then rings at least two extensions at the same time. Not a hunt group, but ringing at the same time as if they were plugged into the same physical port. Does anyone know if this can be done, and if so how? Many thanks, Spencer I know you can do Dial(SIP/101SIP/102) and the like, but you specify you do not want this ( not a hunt group ). How do you want the call to be handled when someone picks up a phone that's ringing? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Steal a call from a SIP extension
Hi all, I think I've seen this somewhere, but I can't remember where; Is it possible to steal a call from a sip extension? Let me explain what we are trying to do: Parking calls is a good thing, but having to remember an extension may be a bit much to ask my user base who is used to seeing line presences on their phones ( old avaya partner ACS system ). I'm thinking they'll keep forgetting what extension a call is parked on. I would like for them to be able to put a call on hold on their extension, and have someone else be able to steal it off that extension from a different extension. Is this possible? If so, can I get the terminology for this ( I can do my own research if needed )? Thanks! Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 phones do not update time from time server
Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. Seshu Can you look for the sntp entry in your ipmid.cfg file and post it in it's entirety? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line Presence:
Hi all With the recent thread on line presence in asterisk, can anybody tell me if there is a phone out there that supports this? Say I have 20 extensions: Is there any way, hardware based, for me to see the activity on those lines. And for a bonus, is there any way for me to interact with them? Thank you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Acceptable voice time delay
chawki, If I may answer this; If you have 600ms round trip to voipjet, I would guess there are further problems with your line than simple latency. That additional 2.5 seconds of delay may be any combination of things, but I would look first to your ISP and their backbone. I have tried voipjet, and while I wasn't enamored with it, I did not find any latency issues that you speak of. Good luck! Sean chawki hammoud wrote: thank you Rob: the problem is that I am experiencing about 3 secs latency although the ping is 600ms which is a round trip packet travel time. so i should experience about half a sec latency including the voipjet server response and the latency to the pstn. that is annoying, but nothing compared to about 3 sec. do you think the rest of the delay is due to voipjet slow response to the pstn network or some other issues would you be bale to clculate where the 3 sec is comming from thanks. Around 250ms max. Over that and you will have the walkie-talkie effect you are experiencing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Acceptable voice time delay
chawki hammoud wrote: What is considered an acceptable time delay between two servers for a fair (not neccessarily great) voice quality. I can't really deal with anything over 150ms, although regular users will tolerate ~200ms. I use voipjet to connect my calls from iax2 to the pstn. Although the sound quality is good, there is considerable time delay, I wait seconds before the other party hear what I say. It becomes more of a walkie talk. When I ping voipjet, it takes about 600ms. There's your problem. 600ms stinks. Thanks a bunch. No problem. I don't know if voicepulse can do Europe iax term, but it's worth looking into. I've had pretty good experiences with them so far ( excepting the price hike...but what can you do? ). Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can you comment on this Qos script? How does one shape RTP?
Honestly, the best script I've ever found is the wondershaper script ( google it ). I tried the correct one posted in this thread, tried modifying it, but in the end I just used wondershaper. Does a great job. My only fear is it doesn't specifically target IAX2 traffic as high priority, but I can modify it later to do so if needed. On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no noticable problems. Along with someone streaming a shoutcast station ( sigh ). The station broke up, but the calls didn't. cmisip wrote: I got this from the voip wiki but the original script didn't seem to work right so I fiddled with it a little bit. I am no expert so maybe someone can look at it for errors. This is for my cable connection. So far asterisk seems to use 1:10 while all other traffic uses 1:102. How does one packet shape RTP? Thanks for any help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [OT]: Wiki Etiquette
Hi folks, I recently registered with the wiki site to fix a few things I've noticed, and I had a question: Is it proper to delete other people's additions if they are obviously incorrect? My main concern is for the content, which is ( well, was ) false. On the other hand, I do not want to start a pissing match with anybody because of bruised egos. Further, in some cases that I've seen, the OP might have a valid point, but it is not one shared by the general populous. In my mind, that view should be respected, but on the other hand, I feel there should be a correction to the wiki regarding it. Any input on this would be greatly apprecaited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] prevent callerid spoofing between asterisks
elkur wrote:
Hi all
For example here is asterisk-A sip.conf:
[123456]
type=friend
secret=xxyyzz
nat=1
qualify=100
host=dynamic
canreinvite=no
callerid= 123456
context=sipphone
dtmfmode=rfc2833
Sip-phone can make and receive calls and callerid
value overrides phones config here, thats nice.
OK, lets set up an Asterisk-B as sip peer and also send all our
outgoing traffic to Asterisk-A, useing same the sip account.
#Asterisk-B sip.conf:
register = 123456:[EMAIL PROTECTED]
[asteriskA]
type=friend
username=123456
secret=xxyyzz
host=ip.adr.of.asteriskA
#Asterisk-B extensions.conf:
exten = _XXX.,1,Dial(SIP/asteriskA/${EXTEN})
Now if I make a call from asterisk-B to asterisk-A, I'm able
to spoof callerid, because in this case asterisk-A doesn't
override callerid. How to prevent that?
Thanks.
Elku
Setup the dialplan on the protected * server to manually set the
callerid. That will over ride any value sent to it by the remote * server.
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Re: [Asterisk-Users] Channel bank question
Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Channel bank question Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean I am assuming you are in the USA, correct me if incorrect. Correct. You want to call your telco and see what the cost of a PRI (T1) is to replace those 10 lines. You have 10 analog lines should be at the point where it is about a break even, if not call a competitive carrier. Not in my area. I have one provider who is brave enough to ATM a t1 out to my location. Everybody else won't touch us. Currently, we have what our vendor is calling a burstable t1. I don't know if this is a common term or not, but bassically it means voice and data share the t1, voice eating into the bandwidth as needed. The t1 is actually terminated into an Adtran 616 which I am currently researching to see if it can feed out a t1 feed instead of the 10 copper lines. But I digress. You do not want to use a channel bank to convert analog to digitial, even if it could be done you are putting bandaids on a huge wound. Agreed. However, given my options You will get a lot of features with the PRI you can not get on analog, not to mention it will work, what you are talking about doing makes no sense from a practical standpoint. Well, except it's probably the best solution when you consider cost/complexity. Do it right, get a PRI and a single PRI digium card (or another PRI terminating device like a T1 chabbel bank)/ Normally, I'd agree with you. However, this situation is different given the line costs. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Petition for IAX firmware
denon wrote: Hi all, I've put together a quick petition, in hopes that we can possibly persuade Sipura (or any other large-scale IP handset manufacturer) to include firmware support for IAX. The IAXy has proven that an IAX product is in demand, and very useful, and I think we'd all like to see a handset manufacturer follow Digium's lead. I'm not particularly endorsing Sipura, however I do know that they have seriously considered support for IAX, and have decided to hold off until the demand is there. I'm hoping that with some numbers, we can prove to them that the demand is already here, and that IAX is already a viable technology. I'd like to encourage everyone to show your support -- hopefully Sipura, and/or other manufacturers will see these hard names and numbers, and realize it's time to move something into production. Petition: http://www.petitiononline.com/IAXPhone Thanks, -d Signed. Sean Kennedy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rookie getting started question
Randy Paries wrote: Hello, I just download and installed the .8 ISO i have installed two Digium Wildcard X100P Cards( I have two outside normal analog lines) Is there a simple of howto to get these working ? I download the handbook, but wow there are many options. I guess, the one question I first should ask, is ?? Should the user be familiar with telephony apps and terms before attempting to use this app. I am a programmer that owns a small business, so all the IT/phone stuff is done by me and that is not my expertise. Thanks Randy The * admin should have some working knowledge, I would suppose. I didn't. I had to bootstrap learn, as it were, just like you are. It's not so bad, once you get the hang of it ( actually, it's quite easy ). If you have specific questions regarding installation, Digium offers free tech support for the initial installation. Or you can ask your specific questions here and we can try to help you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel bank question
Hi all, Quick question regarding channel banks, I managed to confuse myself ( monday...daylight saving time...no coffee ). If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? Thank you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom sound quality problems
Eric Mason wrote: I'm having a problem with my Polycom phones and hoping someone else has experienced the same thing: Outbound calls are fine, and inbound calls originating from another SIP phone are fine, but inbound calls to the Polycom phone from an IAX channel sound like you're talking to a robot. The person on the Polycom sounds fine to the person on the IAX channel, however. Inbound calls to our soft phones sound just fine. Asterisk 1.0.5 on Debian (also had the problem with 1.0 on Fedora) Polycom SoundPoint IP500 SIP Sixtel is the IAX provider. Anyone experience this before or have any ideas? Thanks Eric Check to see what codec is being used for the call. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] username/password for PolyCom IP500 web interface?
Garrett Nelson wrote: Ok, I am still working on getting this PolyCom phone working with Asterisk. I have been looking all over, but I have not been able to find the username and password for the web interface on this phone. I found some site that said it was Polycom and spip, but that does not work. Anyone else have any ideas what it might be? Both PolyCom and the place I bought the phone from are useless for support. -Garrett Polycom/456 Caps are important. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] getting boot server working for PolyCom IP500
Garrett Nelson wrote: Ok, so I got a FTP server set up and dumped the latest firmware in there. I successfully got the phone to connect to the FTP server and upgrade it's software. However, I now want to place a customized sip.conf file on the FTP server so the phone configures itself for Asterisk. From http://www.voip-info.org/wiki-Polycom+Phones : To set these phones up with Asterisk you need to put configuration files based on the phone's MAC address on an FTP server that the phone downloads from. The phone also downloads it's firmware from that same location. The phones can also be manually configured without a boot server but not all features are accessible. So what does that mean exactly? Do I make a folder in the home directory of the FTP server that is named the MAC address of the phone? And then just stick the SIP.conf file in there? I'm slowly figuring this stuff out, thanks for the help everyone. I still can't get into the web interface on the phone, I've pretty much given up on that. -Garrett No, what you do is put a file named MAC ADDRESS.cfg in the ftp directory, and that file will point to the rest of the config files. For example, mine looks something like this: !-- ster SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.10 $ $Date: Jan 29 2003 14:19:22 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone101.cfg,sip-modesto.cfg,ipmid.cfg MISC_FILES= LOG_FILE_DIRECTORY=log/ sip-modesto.sip is the file configuration file in my case, phone101.cfg holds the phone's login settings and ipmid.cfg is the catch all EVERYTHING GOES HERE configuration file that covers all phones. So each phone should have it's own *.cfg file and phonex.cfg file, but everything else is shared. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [newbie question]Can I can from a phone through Asterisk to another Asterisk server to call out from the 2nd Asterisk server
Koa CG wrote: 1. I wonder Asterisk can do this (refer to the following diagram) or not ? (Can I make a call from the SIP phone to the normal phone ) Asteriskinternet Asteriskcall to normal phone/ # Server 1 == Server 2 # normal phone / SIP phone cellphone 2. Is the Asterisk server 2 called the PSTN Gateway ? 3. What are the hardware that I need to do that ? Hope that anyone can help me in this newbie question , thanks in advance for all . Rgds, Koa I am not quite sure what your diagram is trying to convey, but I will just say Probably.If you can get the channel into asterisk, then you can do anything you want with it. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse connect has doubled their rates
Tim Burt wrote: Today I received an email informing me that effective April 1, my per number charge for VOIP will almost double. This is the downside of VOIP. It is unregulated. I have published and distributed my new VOIP phone number, and now, with no warning, my monthly charge has doubled. Ouch.. Beware of which provider you choose! There is nothing to prevent them from doubling my rates again on May 1st! Getting a little dramatic there, aren't we? It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). And it's still cheaper than my land line, when you consider that all incoming calls are free, as well as all 1800 numbers. For everything else, there's voip-jet. Not that I apprecaite the raise much myself, but hey, this industry is still in it's infancy. It gets too bad, someone else will take their place. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse connect has doubled their rates
Max W Blackmer Jr wrote: It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). h, actually it is only a 28% increase. you want to see outrageous you should see my gas bill. My bad. *I* must be using that new math now. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse connect has doubled their rates
Perhaps I misspoke. A land line would run me ~20 bucks a month. A VP number will run me 11 bucks a month. I only specify the free incoming calls because that's a distguishing characteristic of voip DIDs, many places do not give you free incoming. And anyway, if you are a consumer customer, your incoming calls are not free. So there you go. Sean Mark Musone wrote: Call me silly, but arent incoming calls on land lines also free?? -Mark On Wed, 30 Mar 2005 12:12:29 -0800, Sean Kennedy [EMAIL PROTECTED] wrote: Tim Burt wrote: Today I received an email informing me that effective April 1, my per number charge for VOIP will almost double. This is the downside of VOIP. It is unregulated. I have published and distributed my new VOIP phone number, and now, with no warning, my monthly charge has doubled. Ouch.. Beware of which provider you choose! There is nothing to prevent them from doubling my rates again on May 1st! Getting a little dramatic there, aren't we? It's going from 7.95 ( 8 bucks ) a month to 11 bucks, an increase of 72%. That's hardly what I'd call doubling ( unless you're using that new math I've heard so much about ). And it's still cheaper than my land line, when you consider that all incoming calls are free, as well as all 1800 numbers. For everything else, there's voip-jet. Not that I apprecaite the raise much myself, but hey, this industry is still in it's infancy. It gets too bad, someone else will take their place. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avaya Partner ACS system, pre 7.0
Hi all, I've got an old avaya partner acs 7.0 system here. I'd like to add a simple voip bridge so I can hook up our remote offices. From my research, it would seem the pre-7.0 series doesn't have a t1 port, so if I wanted to do this, I would have to feed the avaya system fxs ports from the asterisk box. Does that sound about right? Has anybody ever done this? Does anybody have any experiences they'd like to share in this area? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya Partner ACS system, pre 7.0
C F wrote: It realy depends what you are trying to accomplish, if all you want to do is add more extensions that happen to be offnet using VoIP, then you could just add analog extensions, and use FXO in * and then IP phones in the remote offices. I'll give you more details: We have two offices. Medium sized office ( 20 phones ), and a small office in a different city ( 5 phones ). We have plans to open a business office that will handle all incoming phone calls. This business office will be next door to the medium sized office. So the plan would be two fold: FXO ports for incoming, and FXS ports to the avaya system itself. Honestly, the way this setup is looking, it may be more problematic to band aid the situation as apposed to simply doing a full phone system upgrade. I'd prefer to do that, but I'm making sure there isn't a shortcut so we could reuse our old equipment and avoid the retraining of our staff. Thank you for the input. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP (music on hold)
[EMAIL PROTECTED] wrote: Does IAX support music on hold? It seems only my SIP phones do. Is this correct? As I understand it, once the call is delivered to asterisk, it becomes abstracted into a channel. And you can do anything to one channel that you can do to other channels ( with a few notable exceptions including zap channels ). So it shouldn't make a difference whether it's sip/iax/zap as far as MoH is concerned. What may cause issues is what class of MoH is specified, by default and otherwise. But as I haven't tinkered with that a great deal yet, I can't tell you much beyond that. Good luck Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple outgoing calls through VOIP providers
David Josephson wrote: Trying to get some straight info from the VOIP providers is difficult. Say there's a small Asterisk switch and it's registered with Broadvoice or LiveVOIP or someone. There are a couple of people using the switch, one is on an outgoing call with the VOIP provider. What happens when someone else initiates another outgoing call through that provider on the same SIP registry? Does * know that the SIP account is busy or does it dial out anyway? Does the provider care? Do I establish a call group of SIP accounts like I would of Zap trunks and Dial/g1 ? Depends on the provider. voicepulse allows up to 4 outgoing and 4 incoming connections on their connect service, which is iax2 by the way. Highly recommended for multiple calls. I imagine if I hit that limit, the call will fail and look for the t or i extension and run that. I can only advise that you try it and watch the console. It's usually pretty clear what's happening. You can then script the extension that it's trying to jump to. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp question ( re: compiling )
Steven Critchfield wrote: On Wed, 2005-03-23 at 17:50 -0800, Sean Kennedy wrote: I am trying to compile spandsp on my asterisk server, and it keeps failing out with the following t4.c:38:21: tiffiop.h: No such file or directory In file included from t4.c:41: spandsp/t4.h:62: error: syntax error before TIFF spandsp/t4.h:62: warning: no semicolon at end of struct or union spandsp/t4.h:63: warning: data definition has no type or storage class spandsp/t4.h:64: error: syntax error before '*' token spandsp/t4.h:64: warning: data definition has no type or storage class spandsp/t4.h:87: error: syntax error before '}' token Using my amazing powers of comrehension, I'm getting that I'm missing the TIFF lib. I'm on fc2, so I do a simple yum search tiff. I then installed libtiff-devel, the only thing I didn't have installed, and I get the same error. Is that not the correct lib? The almighty google didn't help much, so I'm sorta stuck not knowing where to go next. You did most of the right work. Follow the link below to get the proper google search string. It returns 3 links for me and the first one I checked had the proper answer for you. http://tinyurl.com/6ec25 Thanks Steve, Yeah, I didn't realized I'd needed those other files. Once I got that, everything else sorta fell into place ( er..well, in a manner of speaking. Had to manually patch the make file, then I had to learn that spaces are not OK in make files. Then something was borked; I couldn't do make install from the asterisk directory, so I had to copy the so files to asterisk's lib directory. Then I had to copy libspandsp.so.0 to /usr/lib instead of where it's isntalled. But other then that:) ) Thanks for the pointer, now to get to testin'. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Issues
Adam Robins wrote: So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern. I have attached an excerpt from the log file at the end of this message. Has anyone else experienced this? Know what is causing it? Has anyone gotten VoicePulse Connect to work with SIP? Hi Admin, I use the connect service from voicepulse ( as I am sure you do, just specifying for future searches ), and I haven't had any of these problems you have mentioned. I do have a problem when the call is connected, there's about half a second of silence about half a second into the call, on every call. I mention it here in case it's related. Honestly, my first instict says this is a firewall problem. Is that at all possible with your setup? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why even have set CallerID option?
Matthew Boehm wrote: Why even have the ability to set callerid name/number if end offices don't honor it? For example, I have a SIP UA registered and in the sip.conf I have: callerid=Mark Mane 2815692712 When that phone makes an outbound local call, asterisk will terminate it on PRI connected to asterisk box to Time Warner. When the called party looks at their caller id display screen it shows the number that is in sip.conf, but does not show the name I have set in the sip.conf; instead it shows our company name (since we own the number). If it is the responsibility of the last end office to do a data-dip and select out the name, then that means I cannot control the callerid name, correct? Close enough, yeah. So I guess that callerid name is only useful for VoIP-VoIP calls that go thru asterisk? -Matthew Yup. Which is actually very helpful for me. My offices are going to have about 50 phones, and the callerid on the phones will be extremely helpful for sip-sip calls. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Words of a user, ... what can I make better?
Ronald Wiplinger wrote: Words of a user, ... what can I make better? Most of the calls had a little delay. People on the other end of the phone said it sounded like cell phone with little stop during the phone. So, it seems voip is not a good quality pstn phone yet. But I wondered my classmate that using Dynasky calling card with very good quality and just like normal phone. bye Ronald Without knowing anything else, I'd say you didn't use enough magic. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio outband bad quality
Pol wrote: I'm using asterisk as a sip client with a sip proxy server... I've made the pertinent extensions and I've configured the sip.conf correctly or I think so.. I'm using x-lite as a client and when I ring to a public telephone through proxy, the arriving sound it's perfect but the sound I send is very bad, they hear me like a robot and distorted. Anyone know what's the problem? Thank you very much. Pol. What codecs are you using? Between xlite and asterisk, and asterisk and the sip server? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spandsp question ( re: compiling )
I am trying to compile spandsp on my asterisk server, and it keeps failing out with the following t4.c:38:21: tiffiop.h: No such file or directory In file included from t4.c:41: spandsp/t4.h:62: error: syntax error before TIFF spandsp/t4.h:62: warning: no semicolon at end of struct or union spandsp/t4.h:63: warning: data definition has no type or storage class spandsp/t4.h:64: error: syntax error before '*' token spandsp/t4.h:64: warning: data definition has no type or storage class spandsp/t4.h:87: error: syntax error before '}' token Using my amazing powers of comrehension, I'm getting that I'm missing the TIFF lib. I'm on fc2, so I do a simple yum search tiff. I then installed libtiff-devel, the only thing I didn't have installed, and I get the same error. Is that not the correct lib? The almighty google didn't help much, so I'm sorta stuck not knowing where to go next. Thanks! Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP service through Asterisk?
Peter Loron wrote: Greetings. I did some digging with Google, the wiki, and on the archives, but didn't find a recent conclusive answer. If this is answered in the wiki or archives somewhere, please point me to it. I'm in the process of setting up an Asterisk box for home use. I've got a X100P card on the way. I've not decided what analog adapter(s) to get yet. The only phone service to hook up is currently POTS. I'm interested in integrating a VoIP provider into the system (using it as a service for inbound and outbound calls). I understand that I can use Broadvoice (BYOD plan), however I'm also considering other providers. Other than Broadvoice, are there any VoIP providers (Vonage, Packet8, etc) that can be hooked into Asterisk directly? I read about a scheme for Packet8 that involved routing it in through an analog connection on a FXO port...I'd rather have something I can connect in directly. Thanks! -Pete Hi Pete, I use the Voicepulse Connect! service, and I don't have any issues with it. It *is* a bit pricy ( ~3 cents a minute, 7.99 a month for an incoming number ), but I get great voice quality, and I have yet to have an instance where I *can't* dial a number. However, for reference, excluding the incoming number charge, I think I've paid 6 bucks over the past three months in call charges. Your milage will vary of course, but one thing to keep in mind: You don't pay for 1-8xx numbers. So for a business, this would be an awsome plan. And yes, they have direct iax connections. Given my relative noob status, I wouldn't bother with anything else. :) Sean ps- I don't know if the other services let you do this, but voicepulse lets you set your own callerid. Which is, for me, a deal breaker. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern Matching?
[EMAIL PROTECTED] wrote:
I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to
be hands on for each new phone number deployed... so I would like to set
up some administrative extensions that can record greetings... lets say:
[admin]
exten = 8(NXXNXX),1,Record($1|-greeting.gsm)
[incoming]
exten = _(NXXNXX),1,Playback($1|-greeting)
exten = _(NXXNXX),2,Goto($1,1000)
exten = _(NXXNXX),102,Playback(generic-greeting)
[21]
exten = 1000,VoiceMail(2)
[310333]
exten = 1000,VoiceMail(3)
The concept here is like the capture buffer in a Perl regex.
So that if admin dialed 821, it would give them the chance to
record the greeting, which would be put in the 21222-greeting.gsm
file.
If someone called 21, it would play the 21-greeting.gsm
file, if it existed, otherwise if it failed, it would play
generic-greeting.gsm. Then it would change context based in the called
number.
Granted, I'm asking for alot here, but is there any way to approximate
this kind of an advanced configuration with Asterisk?
Steve
Not that difficult. A few things you will need:
${EXTEN} is the current extension dialed
goto statement
You can trim crap off your vars using the ${EXTEN:1} notations. In my
example, I am trimming the front digit off the exten var. If I wanted
to be fancy, I could trim x off the front, and only read for n digits
like this: ${EXTEN:x:n}.
At least, I think I could. Perhaps someone with more recent working
knowledge could confirm that?
It's all the in the wiki. When it's up that is. :)
Sean
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Re: [Asterisk-Users] Different codecs for different numbers via same IAX provider; how? Configs included.
C. Tomlinson wrote: Hi, I have been trying to work this out and havent been able to. I have some incoming numbers that come in over IAX, from the same server, and wish to use different codecs for different calls. This doesnt seem to work for incoming either. I cant seem to get any combination of allow/disallow to work. Ideally the following would work: [general] register = XX disallow=all [XXX] ;incoming number I want to use GSM with type= context= secret= allow=gsm [YY] ;incoming number I want to use alaw with type= context=XX secret=X allow=alaw However they only use one codec for both numbers. Am I doing something wrong in iax.conf, I am running stable, or is this something which * doesnt support yet. Thanks C You will have to setup separate iax2 identities on the calling server ( ie: the server delivering the calls to you ) for this to work. Your receiving server has no way to negotiate a codec based on the incoming phone number ( correct me if I am wrong, but I see no way to do that ). Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone ringing and not going to voicemail?
Matt wrote: Hi, I have one phone on my network that just keeps ringing (when I call it) and does not go to voicemail. If the person there is on the phone, and someone calls it they get the busy message, but they never seem to get the 'unavailable' message... instead it will just ring and ring and ring... any ideas? They are setup with a voicemailbox, and it is set to transfer after 15 seconds of ringing. Your extensions.conf file would be helpful in this case. At least the section containing the extension for the problem phone. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel name (and substring)
Thomas Andrews wrote:
How do I get the bit like IAX2/white_phone in extensions.conf eg from
pre-defined variables or variants thereof ?
What I *do* get is strings like this IAX2/[EMAIL PROTECTED]
from ${CHANNEL}, but that's the full channel name.
What am I missing here ?
Thanks,
Thomas
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This should help:http://www.voip-info.org/wiki-Asterisk+variables
Look especially at the substring section.
Sean
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Re: [Asterisk-Users] Undocumented exten syntax?
John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,SetGlobalVar(EMERGENCY=1)
exten = s,n,SetVar(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
Now, I have several questions:
* What is the n priority and how can they use it for several
different items? Don't they need an increasing integer there?
* What is the (checkavail) doing?
* What does s+2 mean?
I've tried looking in docs and the wiki but can't figure it out.
Thanks!
-- John
I have been curious about this as well. I was thinking it may be pseudo
code? It seems easy enough to read, so that may be what it is.
*shrug* I'd like to know the answer to this as well.
Sean
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[Asterisk-Users] Wiki down: Is there another source for documentation?
As the title suggests, I was wondering if there was another source of documentation for Asterisk. Related: If one wanted to contribute to documentation, who would one contact? Thanks! Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bandwidth
Dunc Depends on the codec: http://www.voip-info.org/wiki-Bandwidth+consumption Offhand, I would recommend hanging on to the fax line, and pipe it into the asterisk box as an emergency line. That way, people can still dial 911. But that's just me, others will have better ideas I'm sure. Sean asterisk wrote: Assuming I'm using a VOIP provider of some sort, what kind of bandwidth requirements / line should I expect to have in place? I currently have 8 traditional voice lines, and a FAX line that doubles as my DSL source. Ballpark, what do I need to have in place to move everything to asterisk? Dunc ___ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicepulse silence during conversations
Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off chance this could be something on my end: Asterisk 1.0.0 Connecting to voicepulse via iax, ulaw codec Polycom 500 IP SIP phone, ulaw codec I'll be honest, I don't notice it at all, but my customer does, and I'd like to make them as happy as I can with this system. Also ( I would feel silly making another thread out of this ) what are the common reasons for echo between sip phones going through two different asterisk servers? As in phone - asterisk A - asterisk B - phone. I've been searching for it, but I'm not having much luck. Thank you, any help is greatly apprecaited! Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming calls
Martin Roy wrote: OK I have 12 phone lines connected to 3 digium TDM04B cards on the same server. I must do the following thing : The first 10 lines will be use by one company and the 2 left by another one. For outgoing calls it's quite easy I just create 2 different group and let them dial on a different one. But for incoming calls how can I setup asterisk to answer on the first 10 lines with one message and on line 11 and 12 with another one? If I put the s,1, Answer thing it will answer all 12 lines with the same message... I'm sure it's easy but I just don't know how to do it. Thanks Martin Making a wild guess: Put the two different lines in a different context. You can then treat them to different s extensions. I am assuming zap? That should be possible, but I haven't had to work with zap lines yet ( I've been lucking with the voip providers ;) ) Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound quality - commercial vs. Asterisk
Paul Fielding wrote: So far in my playing with Asterisk I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters (Grandstream 286, Digium IAXy). I've also got a Vonage line, using a Linksys ATA. None of the devices I've connected to my Asterisk server have been able to maintain the same consistent sound quality over a long distance as the Vonage line.Don't get me wrong, the Grandstreams are actually not too bad, but there is still some breakups that can be annoying. Meanwhile the Vonage ATA maintains an almost flawless connection, all the time. I'm assuming (perhaps wrongly?) that the Linksys ATA that Vonage uses is still using SIP with some standardized codec. If that assumption is correct, then how the heck to they manage to get the consistent connection quality? Is it just a matter of the right setting tweaks within Asterisk and/or the SIP devices? I don't think it's a question of Asterisk hardware, since if I connect via local network to the Asterisk server with a SIP device the quality is pretty consistent. It's generally when remotely connecting that I have the inconsistent sound quality. This would lead me to believe that it's a matter of tweaking something to deal with latency or packet dropping issues (?). What has Vonage got figured out that I still need to? Any comments would be appreciated... regards, Paul Likely, you are running into packet queue problems. As I recall, the vonage device goes on the line before anything else, so it can shape the stream to put it's bits first, ensuring it's packets get out in a timely matter ( #1 important thing in voip ). If you were to shape your stream and put your voip bits first, then I think you'd see an improvement in the qualty of service. Granted, I don't know your particular situation, so this could all be guess work. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
