[Asterisk-Users] How to transmit Video
please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . Yahoo! Mail Use Photomail to share photos without annoying attachments.___ --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] Wanted: IAX ATA w/ FXO
Greetings, I''m looking for an IAX ATA w/ an FXO port. Does such a device exist in the market? SH ___ --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] external modem
2006/3/15, Gidean Chan [EMAIL PROTECTED]: Can Asterisk @ home receive incoming call using a external modem? In general, modems can't be used for voip because most of them can't do full duplex. On the other side, an SPA3000 may be cheaper than some good external modems. Some softmodems uses chips that works, but you must choose the right one. See www.voip-info.org. -- Alejandro Vargas ___ --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] How to transmit Video
RAHEEL HASSAN wrote: please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . Yahoo! Mail Hi, i have used eyebeam exten for video. However it is not free. ___ --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] Feedback from VON expo! Info on * HA and Polycom phone!!
Hey group, I just got back from the VON expo. It was insanethere were so many companies there. The #1 thing ***EVERY*** company focused on was convergance - getting all your communication devices to intergrate with eachother. There were some nifty products out there that did some cool stuff :-) Of course Digium/Asterisk was there and I had a list of questions for them. I went by several times asking more and more questions...by the last visit, these guys were running from me because I was driving them nuts :-) Here are all the questions I asked them (this is not word for word...just a summary): Q: What are the plans for HA? A: With a configuration using DNS-SRV and DUNDi, you can create a pretty resiliant setup now. Q: What about failover without losing a call A: IBM has been able to make asterisk do this. However, at this time we are not working on any solution to offer this as part of the program. Q: Do you plan on offering support for other distros for Asterisk Business Edition? A: [uncertain answer] Not really sure...maybe SuSE...not sure Q: When is asterisk going to fully support video? A: Asterisk can complety support video using H.261, H.263 and we recently added support for H.264 Q: What do you recommend as the best solution for HA? I got two different answers for this from two different people there. Both made good sense and are basically what everyone is doing now. Here both approaces are in a nut-shell: Approach 1 (seemed to be the preferred method): Use DNS-SRV lookups for all registrations. This will distribute the calls among the * servers. Next, you configure your servers using regexten and DUNDi. You use regexten to dynamically create the exten = 1234,1,NoOp when a phone registers with that server. Then when a call comes in, you use DUNDi to try to complete the call locally. If the phone is not registered to that server, then do a DUNDi lookup to find the server that the phone is registered to and then pass the call over IAX to that server to take it to the phone. Of course the phones will need to have a short registration expiration so they update frequently because if the server they are registered to crashes, until it re-registered, no server can access it. But by doing this, the phone will re-register to another server and then the next DUNDi lookup will then go to this new server. I asked about the load of having many phones registering frequently and he said it is no big deal at all. He also said it was very important to make sure cache is disabled in DUNDi!!! Each call that is made should result in a new query. This will ensure the calls are not getting old cached info which may no longer be accurate. Approach 2: Use a SER box to handle all registrations. The SER box will take care of distributing the load between the * boxes. You do not use DUNDi or regexten in this case. Just let each * box function on its own. If one of the servers fails, SER will not use it to terminate calls. Sinces the phones are registering to SER, and all incoming calls will be routed to SER, you do not need to worry much about the * boxes. You just need to make sure you have your SER boxes in a cluster to fail-over in the event of failure. Overall theme of the Asterisk stand: selling third-party products. In the there section, Digium had 10 seperate vendors that have teamed with them to sell special programs/products/services that intergrate with Asterisk. One was a call-center program, another was a resellers package, another delt with firewalls and NAT, another for voice recognition, another was Intel (that has partnered with Digium to offer drivers in the ABE for the intel cards), another was some email, fax, chat, presence, etc. kind of box that sits in front of * to combine all these servicesand some others I dont remember. It felt like I was walking into an infomercial! I also spoke with Polycom guys a great deal and asked many questions: Q: Do you plan on offering 10/100/1000 ports on the phones? A: Yes, in the near future Q: Do you plan on offering a standard phone jack for failover purposes? A: No, we have no talks of this. However, I will take this idea to the production development team. Q: What is the services button ever used for? A: This is only operable in the 601 and is used to launch the XML browser. We have partned with many companies to offer you sports, weather, stock, movie ticket info...etc that can be fed directly to the phones screen. Q: What the deal with the limit on the number of people you can monitor for presence? A: There is no limit in the phone. This is an Asterisk limitation. Q: How can you get the name of the person you are calling to appear on the phone instead of their extension? (they had a demo of their phones there and they were doing this!!!) A: You enter
Re: [Asterisk-Users] Re: how to show called name on callingpolycomdisplay
Hey guys, Got some feedback on this from Polycom. See my post Feedback from VON expo! - Gabe - Original Message - From: Noah Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 15, 2006 11:28 AM Subject: [Asterisk-Users] Re: how to show called name on callingpolycomdisplay This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. This is not just a function of the phone. The phone has no idea what the caller id of the receiving end of the call will be. Something would have to tell it after the call was connected. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. You don't have to create some special sip method or do anything within asterisk at all. You could monitor calls from outside asterisk, and use sipsak to send a special sip message back to the phone after a call was connected. Once you sent it to the phone the magic part would be making the phone do something with it. I'm glad to see the asterisk developers are tackling this as it's a nifty little feature and it's way beyond my skills! - Noah ___ --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
[Asterisk-Users] SIP routing over IAX2
Hi All, I have two Asterisks, one on the LAN that handles the internal calls with a PSTN interface and one on the DMZ with a public interface. I would like to route SIP calls from the internal users to the Internet over IAX2 to the DMZ and onwards. All users have soft phones so they would enter sip:[EMAIL PROTECTED] to get a connection. I would like to avoid having number prefixes to dial external SIP phones. Can Asterisk do this? I am relatively new to Asterisk. I guess I'm after something like an email smarthost feature for SIP. I have googled and checked some of the getting started pages but all dial plans deal with number prefixes to route calls. I want to route calls starting with 'sip:' as a prefix. Thanks, Bart... ___ --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] Double-ring tone
Could be the same problem I had with my Aastra - progressinband=no worked for me. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 15 March 2006 18:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double-ring tone Not sure it's that weird :O Douglas Garstang wrote: The phone must have transported you to Australia... :) -Original Message- From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 15, 2006 10:05 AM To: asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Double-ring tone I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on 7-4. TIA. Julian ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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] carry forward uniqueid
Hi all, I have a couple of asterisk servers running. When one asterisk server dials another asterisk server over IAX, i want to match that call in both of the cdr's. How do i make both asterisk servers use the same uniqueid for that call, if this is possible. Or is this a dumb question since it is 'unique'. I just want to match a call on different asterisk servers from the cdr's. thanks, yusuf ___ --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] asteriskathome maximun channels per trunk
I'm using asteriskathome 2.5. I'm using 2 spa3000 for dialing-out. I configured a trunk for each one with maximun channels=1 and an outbound route that includes both trunks. When a second outgoing call is placed, Asterisk tries to place it in the same that is already in use resulting in a busy tone. ¿What can be the problem? -- Alejandro Vargas ___ --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] GUI Web interface
hi i think that the only way to refresh data on page without reloading is to use ajax ___ --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] Asterisk RealTime Question, Please help
Hi All, Thanks for your replies. I need many contexts because I have around 1000 DID's each with 5-10 Extensions. These DID numbers are changed or added very frequently and whenever there is a change I have to change Extensions.conf manually. So please tell me how can I do this dynamically without changing Extensions.conf and help me configure Asterisk. Thanks once again for your help and time, Manoj. Quoting Benchev [EMAIL PROTECTED]: I was able to install Asterisk and Asterisk-addons and use them successfully. But I have a problem now, I have many contexts and it looks like Asterisk is unable to find the context given directly in Mysql DB unless I specify it in Extensions.conf to switch it to RealTime. If I add a new context in Mysql then I have to add it in Extensions.conf and reload extensions whenever I need a new context. Please tell me if there is a way to avoid all this and make Asterisk take contexts directly from Mysql without mentioning that context in Extensions.conf. If this is possible then I can make my Asterisk RealTime actually and modify contexts directly in Mysql. Idea from the wiki: ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given peer who registers or unregisters with ; us. The actual extension is the 'regexten' parameter of the registering ; peer or its name if 'regexten' is not provided. More than one regexten may ; be supplied if they are separated by ''. Patterns may be used in regexten. ; ;regcontext=sipregistrations That means that you should creat a mother context in extensions.conf: [sipregistrations] But first I would try to add a field regcontext along with regexten(which already there) in sip_users table since for the trick to work ... read http://www.voip-info.org/wiki-Asterisk+sip+regcontext OK, that will enable the auto generation of a context but as the new context won't have a switch statement it doesn't help with this problem... I may try writing a default switch if no matching context found type patch. Well, it wont generate a context, it would rather register the extension of the new user under [sipregistrations] And, maybe now is the time to warn that regexten was created to facilitate a sip-user extensions' propagation within an * network; there is a discussion Clustering going on the list, see for details. As for the switch, since context is optional: (switch = Realtime/@realtime_ext) and if left off, RealTime will use the current context, in this case sipregistrations. Means: [sipregistrations] switch = Realtime/@realtime_ext ;realtime_ext or whatever the table name is Ok i'am guessing sans voir here since I don't understand why so many contexts are needed? Hope it helps, Benchev ___ --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] SIP routing over IAX2
2006/3/16, Bart J. Smit [EMAIL PROTECTED]: Can Asterisk do this? I am relatively new to Asterisk. I guess I'm after something like an email smarthost feature for SIP. Yes, Asterisk can do protocol conversion as well as codec conversion. Just configure phones and asterisk to connect correctly (i.e. echo test working) and make sure the audio codecs you are using are compatible or are enableded in asterisk. I.E. One case that will not work: phone or trunk A: protocols supported speex,iBLC. Asterisk: supports speex, iBLC, G711. Phone B: supports G729, G723. In this case, Asterisk should converted one of the codecs supported by B to one of supported by A, but Asterisk can't decode them because you don't installed any codec for G729 nor G723. Cases it will work: if A supports also G729 or G723: in this case, Asterisk don't need to do transcoding, then it does not matter if it has tihs codecs. If you install G729 and/or G723 in Asterisk. In this case, Asterisk can decode the audio and re-encode with speex or iBLC. -- Alejandro Vargas ___ --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] How to transmit Video
RAHEEL HASSAN wrote: please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . I'm using Vizufon CIP-5500. -- Best regards, Bartosz Piec ___ --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] Flash Operator Panel
Hi! Does anyone know how to configure flash operator panel to be able to transfer/hang up calls? I'm trying to set it up, but for me, it only works as a status monitor, because if (for example) I try to drag a phone icon to transfer a call, it ask me to insert the security_code, then I digit the code I wrote in the configuration file, but it doesn't work. Is there something else to configure? Giuseppe ___ --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] SER Asterisk with DID incoming and out going
Hi all I have badly NATed Clients proble with one way Voice After reading some documents people ask me to use STUN Server But still i have some problem with one way Voice I have setup like below iam trying with 2 extensions 1 extention in the same LAN where the * installed 2 extension in different network, NATed IP , 3. both the side iam use SIPURA 4. i have 2 DID from provider 5. i have route them to appropriate extensions Iam able to make calls in and out but the problem where iam setting up server have limited bandwidth So i have installed G729 codec So i want to make RTP so i made setup caninvite=yes since my provider support that option then my NAT Clients have One way Voice problem So after Reading some DOCS SER, should be able to do this Job so SER can be integrated with *, if yes can any one point me to some URL thanks ram ___ --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] echo problem + choppy sound
Look also at AudioFrames setting on your phone. I read that it needs to match 20ms packet size of Asterisk packets and it depends from codec you use. Mimmus ___ --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] asteriskathome maximun channels per trunk
From http://nerdvittles.com/:
Max Channels Bug Remains. A bug has been reported because of a deprecated
command that makes [EMAIL PROTECTED]'s calculation of maximum channels invalid.
To fix it, goto AMP-Maintenance-Config
Edit-extensions.conf-macro-dialout-trunk and comment out line s,7 so that
it looks like this:
;exten = s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})
Then insert a new line s,7 just below it which looks like this:
exten = s,7,GotoIf($[ ${GROUP_COUNT()} ${OUTMAXCHANS_${ARG1}} ]?108)
Then click the Update button and reload Asterisk to activate the change.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alejandro Vargas
Sent: Thursday, March 16, 2006 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] asteriskathome maximun channels per trunk
I'm using asteriskathome 2.5. I'm using 2 spa3000 for
dialing-out. I configured a trunk for each one with maximun
channels=1 and an outbound route that includes both trunks.
When a second outgoing call is placed, Asterisk tries to
place it in the same that is already in use resulting in a
busy tone. ¿What can be the problem?
--
Alejandro Vargas
___
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Re: [Asterisk-Users] Sync Source: Internally clocked
On 03/16/06 04:45 bails said the following: Hi whatever I set the span line to in zaptel.conf ie span=1,0,0,ccs,hdb3,crc4 span=1,1,0,ccs,hdb3,crc4 span=1,2,0,ccs,hdb3,crc4 why are all your spans numbered 1 ? surely they should be numbered 1,2,3,... ? [i'm assuming that there are many spans in your system] -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] SIP routing over IAX2
Thanks Alejandro, I'm sure the codecs are fine, as I can make calls inbound to the LAN Asterisk. Can you tell me which configuration changes I need to make on each Asterisk to route these calls? Bart... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: 16 March 2006 08:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP routing over IAX2 2006/3/16, Bart J. Smit [EMAIL PROTECTED]: Can Asterisk do this? I am relatively new to Asterisk. I guess I'm after something like an email smarthost feature for SIP. Yes, Asterisk can do protocol conversion as well as codec conversion. Just configure phones and asterisk to connect correctly (i.e. echo test working) and make sure the audio codecs you are using are compatible or are enableded in asterisk. I.E. One case that will not work: phone or trunk A: protocols supported speex,iBLC. Asterisk: supports speex, iBLC, G711. Phone B: supports G729, G723. In this case, Asterisk should converted one of the codecs supported by B to one of supported by A, but Asterisk can't decode them because you don't installed any codec for G729 nor G723. Cases it will work: if A supports also G729 or G723: in this case, Asterisk don't need to do transcoding, then it does not matter if it has tihs codecs. If you install G729 and/or G723 in Asterisk. In this case, Asterisk can decode the audio and re-encode with speex or iBLC. -- Alejandro Vargas ___ --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] Double-ring tone
That's in the [general] section of sip.conf, yes ? How would that affect the 7.4 phones ? Wouldn't want to annoy them ;) Julian. Lee Archer wrote: Could be the same problem I had with my Aastra - progressinband=no worked for me. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 15 March 2006 18:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double-ring tone Not sure it's that weird :O Douglas Garstang wrote: The phone must have transported you to Australia... :) -Original Message- From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 15, 2006 10:05 AM To: asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Double-ring tone I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on 7-4. TIA. Julian ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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] Double-ring tone
Why not just set it for the affected extensions in sip.conf? I did it globally and my GXP's didn't mind. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 16 March 2006 09:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double-ring tone That's in the [general] section of sip.conf, yes ? How would that affect the 7.4 phones ? Wouldn't want to annoy them ;) Julian. Lee Archer wrote: Could be the same problem I had with my Aastra - progressinband=no worked for me. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 15 March 2006 18:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double-ring tone Not sure it's that weird :O Douglas Garstang wrote: The phone must have transported you to Australia... :) -Original Message- From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 15, 2006 10:05 AM To: asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Double-ring tone I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on 7-4. TIA. Julian ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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] Server freeze with meetme and sip GSM users and ztdummy
Hi all The first problem I noticed, is that I get very choppy sound, when there are users wich connect to meetme via GSM. Is there a way to force meetme to user aLaw even if the user is connected via gsm? The second problem is that I already had two server freezes just after a gsm user connected to meetme. And known issues? Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ pgpdXH6QM8xhv.pgp Description: PGP signature ___ --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] SER Asterisk with DID incoming and out going
ram wrote: Hi all I have badly NATed Clients proble with one way Voice After reading some documents people ask me to use STUN Server But still i have some problem with one way Voice use stun on dinamic ip :) I have setup like below iam trying with 2 extensions 1 extention in the same LAN where the * installed 2 extension in different network, NATed IP , 3. both the side iam use SIPURA 4. i have 2 DID from provider 5. i have route them to appropriate extensions Iam able to make calls in and out but the problem where iam setting up server have limited bandwidth So i have installed G729 codec So i want to make RTP so i made setup caninvite=yes canreinvite=no nat=yes since my provider support that option then my NAT Clients have One way Voice problem So after Reading some DOCS SER, should be able to do this Job so SER can be integrated with *, if yes can any one point me to some URL thanks ram ___ --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 -- Поздрави, Андрей Сотиров ___ --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] Attended Transfer - transfer timeout, how to change?
Kevin Bockman wrote:
Barry Flanagan wrote:
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when
we do the transfer and dial the new number, it times out after 3 rings
and then the callee is put back to the original agent.
Where can I adjust the timeout which applies to the number we are
transferring to? I have changed the extension for this number to
timeout at 60 seconds, but that seems to make no difference.
There's no need to ask multiple times. I was a couple days behind. The
list is probably a little delayed because of VON.
Sorry, but I was desparate to find the answer and thought the original
had got lost in the ether!
As far as I know, there isn't a variable for this. I made a patch to
change the hard-coded value. I set it to 20sec instead of 15. Adjust
accordingly.
Perfect, thank you very much. This has solved my problem. I will look at
producing a patch to make this a set-able parameter in features.conf and
submit to the bugtracker.
-Barry Flanagan
Kevin
--- res/res_features.c.dist2006-01-14 16:57:54.0 -0700
+++ res/res_features.c2006-01-14 16:58:40.0 -0700
@@ -721,7 +721,7 @@
cid_name = transferer-cid.cid_name;
if (ast_exists_extension(transferer,
transferer_real_context,xferto, 1, cid_num)) {
snprintf(dialstr, sizeof(dialstr), [EMAIL PROTECTED]/n, xferto,
transferer_real_context);
-newchan = ast_feature_request_and_dial(transferer, Local,
ast_best_codec(transferer-nativeformats), dialstr, 15000, outstate,
cid_num, cid_name);
+newchan = ast_feature_request_and_dial(transferer, Local,
ast_best_codec(transferer-nativeformats), dialstr, 2, outstate,
cid_num, cid_name);
ast_indicate(transferer, -1);
if (newchan) {
res = ast_channel_make_compatible(transferer, newchan);
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Re: [Asterisk-Users] Sync Source: Internally clocked
Dinesh Nair wrote: On 03/16/06 04:45 bails said the following: Hi whatever I set the span line to in zaptel.conf ie span=1,0,0,ccs,hdb3,crc4 span=1,1,0,ccs,hdb3,crc4 span=1,2,0,ccs,hdb3,crc4 why are all your spans numbered 1 ? surely they should be numbered 1,2,3,... ? [i'm assuming that there are many spans in your system] no there is only 1 card this was an example of the same span configured differently. i have seen the bug for this but it says closed. We hook this card/box up to 2 Trend Auroura testers yesterday one in NT mode the other sniffing and the amount of sync errors and crc4 errors make me believe the hardware is faulty. A symtom of this was the ability to make incoming calls which were terminated (sync errors green/red) and the inablility to make any outgoing calls. Bails ___ --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] IAX choppy sound
Hi, Does anyone know what would be acceptable RTT. Is 200ms OK? Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Wednesday, March 15, 2006 18:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX choppy sound On Mar 15, 2006, at 6:36 AM, Stojan Sljivic - GDS wrote: Hi, I have downloaded an IAX softphone and tested the connection locally. The sound is perfect. How should I troubleshoot this IAX connection between these two Asterisk servers? Is there some tool that can help in determining the cause of the choppy sound? Your above ping that show 2% packet loss is a good place to start. You shouldn't be losing that much data. ___ --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
[Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)
Hi everyone, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls dont work. Strange. Anyhow I was getting an error: Process_sdp: No compatible codecs! And from the SIP debug I could see that the incoming SIP INVITE was getting a sip response of 488 Unacceptable here from my asterisk server. After doing a bit of searching I determined that this might be the fault of the codecs particularly the G729 codec. So in the peer block that I have for my PSTN provider in my sip conf I specified allow=g729. I called my PSTN geographic number again and was delighted when the incoming calls worked. However when I next went to make an outgoing call (after having added in the allow=g729 line), I got an infinite loop of warnings: WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8) WARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isnt a multiple of 33 or 65 bytes long from RTP After those warnings I thought there might be a problem with the gsm codec so I commented the lines containing allow=gsm and still kept the line allow=g729 because as Ive said already incoming calls wont work otherwise (but outgoing will). This however just gave another warning: WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 4 while native formats is 256 (read/write=64/64). When I comment this line out again I am back to my original situation where outgoing calls work and incoming dont. Has anyone any idea how I can work around this? Many thanks in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ --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
[Fwd: Re: [Asterisk-Users] Sync Source: Internally clocked]
Hi all, I've attached a copy of the debug from the Trend, if anyone cares to look. I'll probablt get more of a response fromt the list than the automated response from digium :( Thanks Bails ---BeginMessage--- Dinesh Nair wrote: On 03/16/06 04:45 bails said the following: Hi whatever I set the span line to in zaptel.conf ie span=1,0,0,ccs,hdb3,crc4 span=1,1,0,ccs,hdb3,crc4 span=1,2,0,ccs,hdb3,crc4 why are all your spans numbered 1 ? surely they should be numbered 1,2,3,... ? [i'm assuming that there are many spans in your system] no there is only 1 card this was an example of the same span configured differently. i have seen the bug for this but it says closed. We hook this card/box up to 2 Trend Auroura testers yesterday one in NT mode the other sniffing and the amount of sync errors and crc4 errors make me believe the hardware is faulty. A symtom of this was the ability to make incoming calls which were terminated (sync errors green/red) and the inablility to make any outgoing calls. Bails ___ --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 ---End Message--- ___ --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] IAX choppy sound
On Thu, March 16, 2006 12:08, Stojan Sljivic - GDS said: Hi, Does anyone know what would be acceptable RTT. Is 200ms OK? Regards, Stojan Sljivic When any of my VPN tunnels get over 100ms I start to get worried! Avg speeds on the tunnels are below 45 ms... I guess it depends on the level of quality you're used to tho! (As well a how far aprt the networks are... Mine are all in the same country...) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] SER Asterisk with DID incoming and out going
Hi thanks for the reply ya the default is NAT=YES only if i keep reinvite=no, the my server b/w consuming lot since i have bottleneck of server bandwidth ram On 3/16/06, Andrei Sotirov [EMAIL PROTECTED] wrote: ram wrote: Hi all I have badly NATed Clients proble with one way Voice After reading some documents people ask me to use STUN Server But still i have some problem with one way Voiceuse stun on dinamic ip :) I have setup like below iam trying with 2 extensions 1 extention in the same LAN where the* installed 2 extension in different network, NATed IP , 3. both the side iam use SIPURA 4. i have 2 DID from provider 5. i have route them to appropriate extensions Iam able to make calls in and out but the problem where iam setting up server have limited bandwidth So i have installed G729 codec So i want to make RTP so i made setup caninvite=yescanreinvite=nonat=yes since my provider support that option then my NAT Clients have One way Voice problem So after Reading some DOCS SER, should be able to do this Job so SER can be integrated with *, if yes can any one point me to some URL thanks ram ___ --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--Поздрави,Андрей Сотиров___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
[Asterisk-Users] open source queue analyzer
browsing the web i don't find any opensource (and free of charge ) software for the web statistic about queues... i've tries queue_stats made from asteriskguru, it is a good tool, and it is free of charge, but it's not open-source :-( i'm considering to develop myself a web application, before that i would ask you if you are interested of this, i would like to activate a sourceforge project the main requirements of the project are: /// realtime - possibility to login/logout from the queue via web interface - monitor the state of the queue (logged in agent/extension, queued calls, ecc) ///statistics - average wait time - average call time - average calls per agent/extension - average calls per hour - average calls per day - average calls per week - average calls per month ///supervisors - define new users that can access to the software - set for each user the operation to do in the queue (login/logout/real time monitor/statistics) now i've realized the firse section, realtime, and i'm using it in my callcenter sice 2 weeks the software use php and mysql o postresql as database (i would add some ajax module for refreshing some data without reloading the page) so, would you like to contribute? what do you think of that? ___ --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] open source queue analyzer
On 13:11, Thu 16 Mar 06, nik600 wrote: browsing the web i don't find any opensource (and free of charge ) software for the web statistic about queues... i've tries queue_stats made from asteriskguru, it is a good tool, and it is free of charge, but it's not open-source :-( i'm considering to develop myself a web application, before that i would ask you if you are interested of this, i would like to activate a sourceforge project the main requirements of the project are: /// realtime - possibility to login/logout from the queue via web interface - monitor the state of the queue (logged in agent/extension, queued calls, ecc) ///statistics - average wait time - average call time - average calls per agent/extension - average calls per hour - average calls per day - average calls per week - average calls per month ///supervisors - define new users that can access to the software - set for each user the operation to do in the queue (login/logout/real time monitor/statistics) now i've realized the firse section, realtime, and i'm using it in my callcenter sice 2 weeks the software use php and mysql o postresql as database (i would add some ajax module for refreshing some data without reloading the page) so, would you like to contribute? what do you think of that? Sounds like a nice project. If it's on sf.net I will for sure checkout the source and see if I can contribute :) I'm a fulltime php/ajax/mysql/postgres developer. Michiel ___ --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] send text to a device
how can I send text directly to a specific device, something like: exten = 103,1,SendTextToDev(SIP/7, hello) ?? I don't think you can send to a particular device, but you can send it to the device calling if it support it. See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SendText hth ___ --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] stop monitor on transfer
snip In the US I think this illegal? Aren't you supposed to have some sort of notification or beeping to indicate a recorded call to the other party? /snip yes. and that is why a lot of times you will hear calls may be monitored for quality control purposes __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!
/snip Nothing complicated really Just a carrier class solution, with advanced custom routing, incoming and outgoing number blocking (at user/company and global level) and whitelisting, findme/followme, user specific pic codes and rate centres based on number dialled, blocking of specific star code prefixed features, different caller ID based on intra company calls, outside calls, calls overriden to use alternate caller id with feature codes, and not to mention it all has to be HA. I'd been doing it in python written AGI scripts interfacing to custom built MySQL tables. Doing stuff this complex in the dial plan would be a nightmare (oh, did I mention a user web interface so that users can make changes themselves?), and Realtime, well it just has too many limitations. Imagine trying to code choosing a specific pic code based on the number prefix in realtime or the dial plan. For example, 1* might be 1123, 1303* might be something else, and 130* might be another pic code. Throw in findme/followme with caller id based routing, multiple numbers per dial (easy in Asterisk but the MySQL tables start to get complex - actually not that easy if you want to dial a local user AND an OffNet user at the SAME time with redundancy).While we're at it, write the application generic enough so that it can handle Queues, voicemail and everything else. /snip How about having some one else come in to help you. It sures seem's like a load. Convince them that having some one else will get things done faster etc. snip My brain hurts. /snip My brain is hurting from just reading it. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] open source queue analyzer
Michiel van Baak wrote: On 13:11, Thu 16 Mar 06, nik600 wrote: browsing the web i don't find any opensource (and free of charge ) software for the web statistic about queues... i've tries queue_stats made from asteriskguru, it is a good tool, and it is free of charge, but it's not open-source :-( i'm considering to develop myself a web application, before that i would ask you if you are interested of this, i would like to activate a sourceforge project the main requirements of the project are: /// realtime - possibility to login/logout from the queue via web interface - monitor the state of the queue (logged in agent/extension, queued calls, ecc) ///statistics - average wait time - average call time - average calls per agent/extension - average calls per hour - average calls per day - average calls per week - average calls per month ///supervisors - define new users that can access to the software - set for each user the operation to do in the queue (login/logout/real time monitor/statistics) now i've realized the firse section, realtime, and i'm using it in my callcenter sice 2 weeks the software use php and mysql o postresql as database (i would add some ajax module for refreshing some data without reloading the page) so, would you like to contribute? what do you think of that? Sounds like a nice project. If it's on sf.net I will for sure checkout the source and see if I can contribute :) I'm a fulltime php/ajax/mysql/postgres developer. Michiel ___ --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 if this becomes a reality, i am happy to contibute financially. This is eactly what i am looking for here, but dont have the skills to complete it. Terry ___ --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] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!
snip It's making you miserable, and YOU are making a lot of us miserable with your incessant and childish whining. /snip 1)I will go out here and defend doug. It has been a lng time since we have heard him whine. Back in the day his emails werent th best and we asked him to change his tone and he did. We are more than happy to help him. That is what we are here for. All of his emails to the list are for help with complex issues that a lot of us dont deal with as often as he does 2)We need a little chuckle some times, and a stress release. I think doug is speaking for a lot of us here. The other day I was setting up a box and was at it for 6 hours straight with no break. By the time I was done I needed a new pack of smokes and the ash tray was full. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] How to configure PSTN lines permissions todifferent extensions ???
OK now your question is starting to make sense. What happens if bosses line is buzy do calls 'rollover' to line 2 and 3? What criteria will define access to the other lines? Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faisal InamSent: Thursday, March 16, 2006 12:12 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to configure PSTN lines permissions todifferent extensions ??? I have 4 telephone lines in the PBX server. One line will be usedby one extension only (i.e. for the boss) for incoming and outgoing.The remaining lines will be shared by all other employees.Some people will be having access to line 1 only. Some have access to line 1 line2 and some have access to line1, line2 and line 3. I will be grateful for ur help. Thanks a lot. Faisal Yahoo! MailUse Photomail to share photos without annoying attachments. ___ --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] Problem with System() command.
Try it with qoutes "mono script.exe" From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nello GaudinoSent: Thursday, March 16, 2006 2:08 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Problem with System() command. Hi, I have an application, script.exe, written under mono framework and for execute them in my linux box I must write in console: mono script.exe The problem is that when I call this application in dialplan with command: exten = 500,1,System(mono script.exe) the application not run! Somebody can help me to find the problem? 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
[Asterisk-Users] PRI Setup
Hi, Need help from you guys. I had my Asterisk Set-up using PRI card TE110p, and everything working ok. However, I had bad experience with Asterisk answering call The problem was, when outsider calling into Asterisk... Asterisk answered call... CLI Accepting Overlap call from (CALLERIDNUM) to (unspecified) channel 0/31 CLI Starting Simple Switch on Zap31/1 Asterisk wait for 3sec Then jump to the context, and my phone rings. Can some one advice that, reducing the number of sec before passes it to the next context or task??? ___ --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] PRI Setup
Without your configs it ill be hard to see what is going on. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chan (Alpha Trilogies Networks) Sent: Thursday, March 16, 2006 8:15 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PRI Setup Hi, Need help from you guys. I had my Asterisk Set-up using PRI card TE110p, and everything working ok. However, I had bad experience with Asterisk answering call The problem was, when outsider calling into Asterisk... Asterisk answered call... CLI Accepting Overlap call from (CALLERIDNUM) to (unspecified) channel 0/31 CLI Starting Simple Switch on Zap31/1 Asterisk wait for 3sec Then jump to the context, and my phone rings. Can some one advice that, reducing the number of sec before passes it to the next context or task??? ___ --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] Problem with System() command.
Also be aware Asterisk is probably runing in its own, non-root account. It needs execute access to the program, and you need to specify full path. At least thats what worked for me J - dialing 500 on my box does System(/sbin/reboot) ! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nello Gaudino Sent: Thursday, March 16, 2006 9:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with System() command. Hi, I have an application, script.exe, written under mono framework and for execute them in my linux box I must write in console: mono script.exe The problem is that when I call this application in dialplan with command: exten = 500,1,System(mono script.exe) the application not run! Somebody can help me to find the problem? 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] How to transmit Video
Look Eyebeamof Xterm. -Mensaje original-De: RAHEEL HASSAN [mailto:[EMAIL PROTECTED]Enviado el: Thursday, March 16, 2006 4:05 AMPara: asterisk-users@lists.digium.comAsunto: [Asterisk-Users] How to transmit Videoplease tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . Yahoo! MailUse Photomail to share photos without annoying attachments. ___ --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] Do Not Disturb?
Brian McEntire wrote:
I looked on the voip-info wiki and found sparse and conflicting
information on how to do this with Asterisk...
My incoming lines are all on Zaptel. Is there a simple why to
implement a '*363 (do not disturb) toggle via the dialplan?
I have an extension for activating/de-activating voicemail call backs.
You should be able to do the same for DND:
[callback-activate]
exten = 80*,1,Set(CALLBACK=${DB(vmcallback/${CALLERIDNUM})})
exten = 80*,2,GotoIf($[${CALLBACK} = YES]?80*,3:80*,101)
exten = 80*,3,Set(DB(vmcallback/${CALLERIDNUM})=NO)
exten = 80*,4,Playback(local/stutter)
exten = 80*,5,Playback(local/deactivated)
exten = 80*,6,Hangup()
exten = 80*,101,Set(DB(vmcallback/${CALLERIDNUM})=YES)
exten = 80*,102,Playback(local/stutter)
exten = 80*,103,Playback(local/activated)
exten = 80*,104,Hangup()
You can do the same thing with DND. Turn the value on or off, then in
your dial string, check the database value and act accordingly.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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[Asterisk-Users] can't get TDM400P to answer
Hi all, I can't figure out why my TDM400P (with one FXO plugin) won't answer any calls. There are no messages in the Asterisk console when a call is placed to the FXO line from the PSTN. Any suggestions would be most appreciated. The wctdm and zaptel modules are loaded: [EMAIL PROTECTED] asterisk]# lsmod | grep wc wctdm 37952 0 zaptel189700 1 wctdm The green LED on the input connector is lit. ztcfg says: ... Channel map: Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. My zaptel.conf is: fxsks=4 loadzone=us defaultzone=us and zapata.conf is: [channels] group=1 context=tdm400p-inbound signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 The relevant sections of extensions.conf are: [tdm400p-inbound] exten = s,1,Ringing() exten = s,2,Goto(MainMenu,s,1) exten = s,3,Hangup; ... [MainMenu] exten = s,1,Wait,3 ; ring for 1 second exten = s,2,Answer ; answer exten = s,3,Background,welcome etc... The TDM400P power connector is attached, even though it isn't supposed to be required for FXO modules. Any ideas? - Mike ___ --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] cisco 7912 not taking config
Jerry Geis wrote: however all the phone shows is the initial config from my office. Its either not picking it up, rejecting it or something??? Doing a diff between the txt files from my office and the second location shows only the proxy and UID and password fields as being different. Just a guess. You didn't have the phone restart in the office so it read the new config. You can have them change the settings via the keypad though. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Flash Operator Panel
Yes, after putting in the code drag it again. Also what does your op_panel.cfg look like? FOP has it's own mailing list, you should try it there. On 3/16/06, Giuseppe [EMAIL PROTECTED] wrote: Hi! Does anyone know how to configure flash operator panel to be able to transfer/hang up calls? I'm trying to set it up, but for me, it only works as a status monitor, because if (for example) I try to drag a phone icon to transfer a call, it ask me to insert the security_code, then I digit the code I wrote in the configuration file, but it doesn't work. Is there something else to configure? Giuseppe ___ --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] Echo canceller data-points
Here is the patch file which I use (I manually removed some other parts of the patch, so I hope it is okay!) - It should be sufficient to get you going. cd into the zaptel-1.0.9.2 source directory, and patch -p1 zap-patch.txt Cheers, Steve On 3/15/06, Colin Anderson [EMAIL PROTECTED] wrote: Is it onerous to backport or is it a case of fiddling around with the makefile? Care to post a backported tar? -Original Message- From: Steve Davies [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 15, 2006 2:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Echo canceller data-points In case this is useful to someone... Initially running * 1.0.7 and the default canceller, about 1 in 20 E1 PRI calls still had echo, sometimes quite bad. Updated to * 1.0.9, and backported KB1 from 1.2 HEAD to this build - No noticable change over the previous version, but we ran with it anyway as the small changes in the EC code looked sensible. Recently we backported MG2 from Zaptel 1.2.4 into our * 1.0.9 build, and noticed a significant improvement. I thought I would try the 1.2 trunk/HEAD version of MG2 with the extra knobs and whistles, but found 2 problems. This version trains even a normally clean line in about 10 seconds, rather than instantly, and its CPU usage is through the roof compared to the 1.2.4 version of the code. (FYI I got very similar resuls at all intermediate SVN versions of the MG2 canceller between 1.2.4 and HEAD) My advice: Go with the 1.2.4 MG2 echo canceller, perhaps if you have plenty of spare CPU the newer code will become useful, but I could not cancel even 20 PRI channels using a 1GHz processor on the latest code - I got clicks, buzzing and eventually a dead PRI. With the 1.2.4 branch I had 40% CPU free when cancelling 30 channels. Hope this helps. Cheers, Steve zap-patch.txt.gz Description: GNU Zip compressed data ___ --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] (unexplicable) peaks of machine load
Matt Florell ha scritto: I've noticed this as well from pre 1.0 versions through to 1.2.5 across 12 separate Asterisk servers. The severity seems to be random mostly. I still haven't figured out what is causing it. MATT--- Your file system is journaled ? this is another common thing that came to my mind (ext3) On 3/15/06, Simone Cittadini [EMAIL PROTECTED] wrote: I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other service and a two line extension. I thought it can be a problem with digium cards but the interrupts aren't shared, and I have the same problem on a pure-voip server. Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6 (right ones for the installed cpu, not generic 386) The only things in common are : Linux debian, iax channels are used, with jitterbuffer ___ --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 - calls going to agents lised as In use
Grretings to all, I am having a problem with a customer's queue setup that I don't really understand. Background: Customer has 5+ queues and is using dynamic login to the queues based on SIP/XXX for example. There is a litle script that runs that allows agents to log into particular queues via the keypad. The user can log in to any queue that he wants, including multiple queues. The customer is using Cisco 7940 SIP firmware. Everything works as expected except the following. When an agent is logged into a queue (one or more), this agent will still get calls from this queue or even any other queue that the agent is logged into) even though the agent is shown as In Use by Asterisk. The Agent gets a call waiting beep. I know that call waiting could be disabled on the phones, but this is not what we want to do. The agent now just ignores the call waiting beep,and continues working. I am wondering if this same problem would occur if we were using agents instead of real extensions. Has anyone come up with a solution for this, or know if Asterisk properly treats Agents who are In use. Regards to all. Joe ___ --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] Re: transfers/parked calls + polycom 501
I am not sure what I did but blind transfers do not work. The Polycom does not allow me to dial the extension of the person I want to transfer to after I hit:transfer - blindthanks "Mojo with Horan Company, LLC" [EMAIL PROTECTED] wrote: When you hit the polycom's transfer button, a softkey appears on the screen that says "Blind" -- hitting this changes the transfer from attended to blind, and the blind button then disappears to show this. There's no real way I know to make this permanent.Andrew Kohlsmith wrote: On Monday 13 March 2006 10:20, Noah Miller wrote: The transer button on the polycom phone does not seem to transfer/park the call properly. I have to use the # - 700 to park the call. If I recall, us ing the Polycom transfer, you have to make sure it is done as a blind transfer. The Polycom attended transfer (default) option does not work. How is this configured? That is, how do I configure the Polycom's transfer button to be a blind transfer? -A. ___ --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 -- Mojo Office Manger, Horan Company, LLC(907) 747- x112___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail Use Photomail to share photos without annoying attachments.___ --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] Echo canceller data-points
oops. attachments are blocked :) I'll email it directly to anyone who provides an email address. Regards, Steve On 3/16/06, Steve Davies [EMAIL PROTECTED] wrote: Here is the patch file which I use (I manually removed some other parts of the patch, so I hope it is okay!) - It should be sufficient to get you going. cd into the zaptel-1.0.9.2 source directory, and patch -p1 zap-patch.txt Cheers, Steve On 3/15/06, Colin Anderson [EMAIL PROTECTED] wrote: Is it onerous to backport or is it a case of fiddling around with the makefile? Care to post a backported tar? -Original Message- From: Steve Davies [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 15, 2006 2:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Echo canceller data-points In case this is useful to someone... Initially running * 1.0.7 and the default canceller, about 1 in 20 E1 PRI calls still had echo, sometimes quite bad. Updated to * 1.0.9, and backported KB1 from 1.2 HEAD to this build - No noticable change over the previous version, but we ran with it anyway as the small changes in the EC code looked sensible. Recently we backported MG2 from Zaptel 1.2.4 into our * 1.0.9 build, and noticed a significant improvement. I thought I would try the 1.2 trunk/HEAD version of MG2 with the extra knobs and whistles, but found 2 problems. This version trains even a normally clean line in about 10 seconds, rather than instantly, and its CPU usage is through the roof compared to the 1.2.4 version of the code. (FYI I got very similar resuls at all intermediate SVN versions of the MG2 canceller between 1.2.4 and HEAD) My advice: Go with the 1.2.4 MG2 echo canceller, perhaps if you have plenty of spare CPU the newer code will become useful, but I could not cancel even 20 PRI channels using a 1GHz processor on the latest code - I got clicks, buzzing and eventually a dead PRI. With the 1.2.4 branch I had 40% CPU free when cancelling 30 channels. Hope this helps. Cheers, Steve ___ --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] (unexplicable) peaks of machine load
Yep I use ext3, have you run test with any other file system? MATT--- Your file system is journaled ? this is another common thing that came to my mind (ext3) ___ --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] Feedback from VON expo! Info on * HA and Polycomphone!!
Gabe. Who was the call-center program from? Thanks, Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Afana Sent: Thursday, March 16, 2006 2:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!! Hey group, I just got back from the VON expo. It was insanethere were so many companies there. The #1 thing ***EVERY*** company focused on was convergance - getting all your communication devices to intergrate with eachother. There were some nifty products out there that did some cool stuff :-) Of course Digium/Asterisk was there and I had a list of questions for them. I went by several times asking more and more questions...by the last visit, these guys were running from me because I was driving them nuts :-) Here are all the questions I asked them (this is not word for word...just a summary): Q: What are the plans for HA? A: With a configuration using DNS-SRV and DUNDi, you can create a pretty resiliant setup now. Q: What about failover without losing a call A: IBM has been able to make asterisk do this. However, at this time we are not working on any solution to offer this as part of the program. Q: Do you plan on offering support for other distros for Asterisk Business Edition? A: [uncertain answer] Not really sure...maybe SuSE...not sure Q: When is asterisk going to fully support video? A: Asterisk can complety support video using H.261, H.263 and we recently added support for H.264 Q: What do you recommend as the best solution for HA? I got two different answers for this from two different people there. Both made good sense and are basically what everyone is doing now. Here both approaces are in a nut-shell: Approach 1 (seemed to be the preferred method): Use DNS-SRV lookups for all registrations. This will distribute the calls among the * servers. Next, you configure your servers using regexten and DUNDi. You use regexten to dynamically create the exten = 1234,1,NoOp when a phone registers with that server. Then when a call comes in, you use DUNDi to try to complete the call locally. If the phone is not registered to that server, then do a DUNDi lookup to find the server that the phone is registered to and then pass the call over IAX to that server to take it to the phone. Of course the phones will need to have a short registration expiration so they update frequently because if the server they are registered to crashes, until it re-registered, no server can access it. But by doing this, the phone will re-register to another server and then the next DUNDi lookup will then go to this new server. I asked about the load of having many phones registering frequently and he said it is no big deal at all. He also said it was very important to make sure cache is disabled in DUNDi!!! Each call that is made should result in a new query. This will ensure the calls are not getting old cached info which may no longer be accurate. Approach 2: Use a SER box to handle all registrations. The SER box will take care of distributing the load between the * boxes. You do not use DUNDi or regexten in this case. Just let each * box function on its own. If one of the servers fails, SER will not use it to terminate calls. Sinces the phones are registering to SER, and all incoming calls will be routed to SER, you do not need to worry much about the * boxes. You just need to make sure you have your SER boxes in a cluster to fail-over in the event of failure. Overall theme of the Asterisk stand: selling third-party products. In the there section, Digium had 10 seperate vendors that have teamed with them to sell special programs/products/services that intergrate with Asterisk. One was a call-center program, another was a resellers package, another delt with firewalls and NAT, another for voice recognition, another was Intel (that has partnered with Digium to offer drivers in the ABE for the intel cards), another was some email, fax, chat, presence, etc. kind of box that sits in front of * to combine all these servicesand some others I dont remember. It felt like I was walking into an infomercial! I also spoke with Polycom guys a great deal and asked many questions: Q: Do you plan on offering 10/100/1000 ports on the phones? A: Yes, in the near future Q: Do you plan on offering a standard phone jack for failover purposes? A: No, we have no talks of this. However, I will take this idea to the production development team. Q: What is the services button ever used for? A: This is only operable in the 601 and is used to launch the XML browser. We have partned with many companies to offer you sports, weather, stock, movie ticket info...etc that can be fed directly to the phones screen. Q:
[Asterisk-Users] ISDN BRI and UK Premium Rate Numbers
Can anyone help point me in the right direction please? I'm based in the UK and I want to start using a Premium Rate number with Asterisk - I think the equivalent in the US would be a 900 number. Effectively the caller pays much more to call such a number than a normal national or local call. The problem with these is that I don't want Asterisk to actually signal to the telephone network that the call has been answered until someone really does answer it, otherwise the caller will be paying a premium rate just to listen to an Asterisk-generated ring tone until someone answers the call. My setup would be chan_capi-cm and an ISDN BRI line with several MSNs (not DDIs -- this line does not support point-to-point only point to multipoint but we do have another line that does do point to point and has DDIs, and if necessary we can use it), and of course Asterisk and various SIP phones. I have very little idea where to start, as everything I normally do with Asterisk involves the call being answered immediately then put in a queue, which is no good in this case. What I really want is for the call to come in then: 1) One or more SIP phones will ring (unless they are on a call) but for Asterisk not to signal an answer just yet 2) Only when someone is free and answers the call does asterisk answer and put them through. Ideally I'd also like the caller and the person answering the call to hear a recorded message saying that calls to this number cost X per minute ... blah blah, this message being triggered only when someone answers the call. This will warn the caller *and* the person answering that this is a premium-rate call. The person answering the call will know to speak after this message has been played. But that's just an ideal situation. Right now I'm more concerned about how to stop Asterisk answering until someone is available to take the call. Can anyone help please? I don't really know where to start. The Wiki seems to be pointing me towards using DID/DDIs, but that's about as far as I've got. NOTE: We don't need the actual Premium Rate numbers themselves. We have those already (we used them with an old telephone system until recently). My problem is just to get Asterisk to work with them in the way I've outlined. Faris. ___ --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] How to transmit Video
Hi You could use windows messenger, kapanga or sipps (deppends on what you want) Jose Manuel Cortes David X Semestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD JAVERIANA De: [EMAIL PROTECTED] en nombre de RAHEEL HASSAN Enviado el: Jue 16/03/2006 3:05 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] How to transmit Video please tell me that what sip based softphone will beused with Asterisk so that i can trasmit and receive video on my LAN . Yahoo! Mail Use Photomail http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=38867/*http://photomail.mail.yahoo.com to share photos without annoying attachments. winmail.dat___ --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] Re: did from sip trunk
2006/2/22, Alejandro Vargas [EMAIL PROTECTED]: I want to do inbound routing of calls comming from sip trunks. Is there a way to force the DID that comes from a trunk that does not have DID support? (something like using the outgoing caller-id for the trunk?) I answers myself. To identify from where is camming the call, I configured SPA3000 dialplan for inbound calls like this: (S0:[EMAIL PROTECTED]:5060) Then, Asterisk sees line2 as DID and I can make inbound routing with this. -- Alejandro Vargas ___ --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] Problems with installing a TE110P on a Dell Poweredge 850 running Fedora Core 4
Hi John, I had the same error when configuring our TE110P. The only way I was able to fix this error was to physically move the card to a different PCI slot. Please note the server I used was the IBM x206 server. Hope this is of some use. Cheers, Phil. John Fulton [EMAIL PROTECTED] etTo Sent by: asterisk-users@lists.digium.com asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [Asterisk-Users] Problems with installing a TE110P on a Dell 15/03/2006 00:32 Poweredge 850 running Fedora Core 4 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com When running a modprobe with a -i to prevent ztcfg from running, the modules install fine: -- [EMAIL PROTECTED] ~]# modprobe -vi wct1xxp insmod /lib/modules/2.6.15i686-smp-TelAK-1.00/kernel/lib/crc-ccitt.ko insmod /lib/modules/2.6.15i686-smp-TelAK-1.00/misc/zaptel.ko insmod /lib/modules/2.6.15i686-smp-TelAK-1.00/misc/wct1xxp.ko [EMAIL PROTECTED] ~]# -- Now, when I run ztcfg, I get the following: -- [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Groundstart (Default) (Slaves: 01) 1 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) [EMAIL PROTECTED] ~]# -- My configs are pretty simple for initial setup: /etc/zaptel.conf -- span=1,0,0,esf,b8zs fxsgs=1 loadzone=us defaultzone=us -- /etc/asterisk/zapata.conf -- [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no -- output of lspci: -- 00:00.0 Host bridge: Intel Corporation E7230 Memory Controller Hub 00:01.0 PCI bridge: Intel Corporation E7230 PCI Express Root Port 00:1c.0 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 1 (rev 01) 00:1c.4 PCI bridge: Intel Corporation 82801GR/GH/GHM (ICH7 Family) PCI Express Port 5 (rev 01) 00:1c.5 PCI bridge: Intel Corporation 82801GR/GH/GHM (ICH7 Family) PCI Express Port 6 (rev 01) 00:1d.0 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #1 (rev 01) 00:1d.1 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #2 (rev 01) 00:1d.2 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #3 (rev 01) 00:1d.7 USB Controller: Intel Corporation 82801G (ICH7 Family) USB2 EHCI Controller (rev 01) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1) 00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01) 00:1f.2 IDE interface: Intel Corporation 82801GB/GR/GH (ICH7 Family) Serial ATA Storage Controllers cc=IDE (rev 01) 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 01) 02:00.0 PCI bridge: Intel Corporation 6702PXH PCI Express-to-PCI Bridge A (rev 09) 03:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet
Re: [Asterisk-Users] (unexplicable) peaks of machine load
Matt Florell ha scritto: Yep I use ext3, have you run test with any other file system? MATT--- No, I will do when I have time (and a server to test on) Your file system is journaled ? this is another common thing that came to my mind (ext3) ___ --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] module load order for Junghanns qozap and TDM card
Hi all, I'm trying to get a junghanns QuadBRI to coexist in the same machine as a Digium TDM400P card (so I can run the ISDN lines in and bridge with analog phones plugged into the TDM). I'm having a problem loading the modules. If I follow the BRIstuff (0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod qozap.o I'm on Debian 2.4.31. That works. But then I still need the Digium module. (modprobe wctdm) I've tried a few different orders. Sometimes I can get the digium to load, and the qozap. but then I get an error on the ztcfg about Span invalid argument (could be my zaptel.conf I realize...) *If* I try loading the wctdm after the zaptel and qozap, the server freezes! Some loop about qozap - dropped audio card I don't know if the quadBRI and the TDM are conflicting/sharing the zaptel module, or if I need to modprobe zaptel before each of them? and in what order? Any suggestions appreciated... I haven't even got to figuring out what I can do with chan_capi, just want to get the BRI card on and stuff. Thanks for any ideas! -- Chris Earle -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] ODBC voicemail storage
Anyone using ODBC voicemail storage in mySQL? For what volume of voicemail? Any performance issues? Seems like a key piece of the failover clustering puzzle (vs. syncing file systems). ___ --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] Feedback from VON expo! Info on * HA andPolycomphone!!
Great Email. I'm going to respond to some of the points. Q: What are the plans for HA? A: With a configuration using DNS-SRV and DUNDi, you can create a pretty resiliant setup now. That's BS. Last time I checked, Asterisk's support of SRV was to only grab the first SRV entry. Period. If it doesn't try any more SRV hosts after the first fails, just exactly how is that redundant? Q: What the deal with the limit on the number of people you can monitor for presence? A: There is no limit in the phone. This is an Asterisk limitation. That's BS too. I have an email thread from a Polycom employee where they recognised it was a Polycom issue and was told they might have an newer version of the SIP software out to address this by summer. Still can't fathom why this takes months to fix, but anyway... Q: Whats the best way to program the phone to handle failover? A: Use a DNS-SRV address for the primary server. When the phone queries the DNS server, it will receive a list of all the possible servers This is broken to some degree. When the phone refreshes it's cache, and grabs the list of SRV servers again, it will continue to use them in the same manner until it refreshes it's cache again, or there is a failure, even when all SRV hosts have the same priority and weight. It should round robin in this case. And in regards to Asterisk HA, and approach #2. If you have your SER boxes use the send() command to stateless forward registrations, you can send registrations from the phones to ALL your Asterisk systems so that every Asterisk box knows about every phone, and every Asterisk box can route calls from/to any phone. Doug -Original Message- From: Jim Houser [mailto:[EMAIL PROTECTED] Sent: Thursday, March 16, 2006 7:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!! Gabe. Who was the call-center program from? Thanks, Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Afana Sent: Thursday, March 16, 2006 2:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!! Hey group, I just got back from the VON expo. It was insanethere were so many companies there. The #1 thing ***EVERY*** company focused on was convergance - getting all your communication devices to intergrate with eachother. There were some nifty products out there that did some cool stuff :-) Of course Digium/Asterisk was there and I had a list of questions for them. I went by several times asking more and more questions...by the last visit, these guys were running from me because I was driving them nuts :-) Here are all the questions I asked them (this is not word for word...just a summary): Q: What are the plans for HA? A: With a configuration using DNS-SRV and DUNDi, you can create a pretty resiliant setup now. Q: What about failover without losing a call A: IBM has been able to make asterisk do this. However, at this time we are not working on any solution to offer this as part of the program. Q: Do you plan on offering support for other distros for Asterisk Business Edition? A: [uncertain answer] Not really sure...maybe SuSE...not sure Q: When is asterisk going to fully support video? A: Asterisk can complety support video using H.261, H.263 and we recently added support for H.264 Q: What do you recommend as the best solution for HA? I got two different answers for this from two different people there. Both made good sense and are basically what everyone is doing now. Here both approaces are in a nut-shell: Approach 1 (seemed to be the preferred method): Use DNS-SRV lookups for all registrations. This will distribute the calls among the * servers. Next, you configure your servers using regexten and DUNDi. You use regexten to dynamically create the exten = 1234,1,NoOp when a phone registers with that server. Then when a call comes in, you use DUNDi to try to complete the call locally. If the phone is not registered to that server, then do a DUNDi lookup to find the server that the phone is registered to and then pass the call over IAX to that server to take it to the phone. Of course the phones will need to have a short registration expiration so they update frequently because if the server they are registered to crashes, until it re-registered, no server can access it. But by doing this, the phone will re-register to another server and then the next DUNDi lookup will then go to this new server. I asked about the load of having many phones registering frequently and he said it is no big deal at all. He also said it was very important to make sure cache is disabled in DUNDi!!! Each call that is made should result in a new query. This will
[Asterisk-Users] Attended call transfer with GXP-2000
Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus ___ --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] Feedback from VON expo! Info on * HAandPolycomphone!!
Q: What are the plans for HA? That's BS. Last time I checked, Asterisk's support of SRV was to only grab the first SRV entry. Period. If it doesn't try any more SRV hosts after the first fails, just exactly how is that redundant? This is for the phones to fail over NOT Asterisk, remember in this case Asterisk has died so no matter what order it 'resolves' it doesn't mater in this case. Q: What the deal with the limit on the number of people you can monitor for presence? A: There is no limit in the phone. This is an Asterisk limitation. That's BS too. I have an email thread from a Polycom employee where they recognised it was a Polycom issue and was told they might have an newer version of the SIP software out to address this by summer. Still can't fathom why this takes months to fix, but anyway... Can you post a referance to the tread? Q: Whats the best way to program the phone to handle failover? A: Use a DNS-SRV address for the primary server. When the phone queries the DNS server, it will receive a list of all the possible servers This is broken to some degree. When the phone refreshes it's cache, and grabs the list of SRV servers again, it will continue to use them in the same manner until it refreshes it's cache again, or there is a failure, even when all SRV hosts have the same priority and weight. It should round robin in this case. Agreed. And in regards to Asterisk HA, and approach #2. If you have your SER boxes use the send() command to stateless forward registrations, you can send registrations from the phones to ALL your Asterisk systems so that every Asterisk box knows about every phone, and every Asterisk box can route calls from/to any phone. Then you have issues with hints, voicemail, and other features. I will concur with you that at this time there is no simple and quick solution to HA on *. It is what it is. I think that we are still in the womb when it comes to VoIP. Phones have become many things to many people, we have to realize that it has taken 50+ years for the phone to evolve into what it is today. Many of the features we take for granted today, (911, callerID, VoiceMail, Echo Cancelation) have only really matured in the past 15-20 years. We got a long way to go.. ___ --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] Attended call transfer with GXP-2000
If you have Line 1 on hold, and you on a call on Line 2, then hitting TRNF and hitting Line 1 will transfer Line 2 to Line 1. Same concept as Conference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus Sent: Thursday, March 16, 2006 7:30 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Attended call transfer with GXP-2000 Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus ___ --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
[Asterisk-Users] Fw: help required configuring card
- Original Message - From: rnacharya To: asterisk-users@lists.digium.com Sent: Wednesday, March 15, 2006 5:56 AM Subject: help required configuring card Hii, I've a Digium TE205P card and I'm running [EMAIL PROTECTED] in a box.I want to configure this card in that box.But as it is a dual port card I'm not sure how to configure it in my box.Basically I want to connect one T1 connection in one port and in anather port of the card I'll attach my Epbx so that I can make use of existing telephone.If any body can suggest what is the cable type required for connecting the EPBX to the card.Please help.The protocol in T1 we are using is EM.regardsrudra ___ --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] Feedback from VON expo! Info on *HAandPolycomphone!!
-Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Thursday, March 16, 2006 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on *HAandPolycomphone!! Q: What are the plans for HA? That's BS. Last time I checked, Asterisk's support of SRV was to only grab the first SRV entry. Period. If it doesn't try any more SRV hosts after the first fails, just exactly how is that redundant? This is for the phones to fail over NOT Asterisk, remember in this case Asterisk has died so no matter what order it 'resolves' it doesn't mater in this case. I disagree. Our Asterisk boxes talk to a proxy server in certain situations. If those proxy servers where in a domain as SRV records, and one of them failed, Asterisk should try each of them in an order defined by the priority and weight. Q: What the deal with the limit on the number of people you can monitor for presence? A: There is no limit in the phone. This is an Asterisk limitation. That's BS too. I have an email thread from a Polycom employee where they recognised it was a Polycom issue and was told they might have an newer version of the SIP software out to address this by summer. Still can't fathom why this takes months to fix, but anyway... Can you post a referance to the tread? Well it's an email thread. I'll forward it to you by email. Q: Whats the best way to program the phone to handle failover? A: Use a DNS-SRV address for the primary server. When the phone queries the DNS server, it will receive a list of all the possible servers This is broken to some degree. When the phone refreshes it's cache, and grabs the list of SRV servers again, it will continue to use them in the same manner until it refreshes it's cache again, or there is a failure, even when all SRV hosts have the same priority and weight. It should round robin in this case. Agreed. And in regards to Asterisk HA, and approach #2. If you have your SER boxes use the send() command to stateless forward registrations, you can send registrations from the phones to ALL your Asterisk systems so that every Asterisk box knows about every phone, and every Asterisk box can route calls from/to any phone. Then you have issues with hints, voicemail, and other features. Hints, voicemail and other features, to this point, are all working fine. The OpenSER systems routes SUBSCRIBE/NOTIFY/MESSAGE etc messages to /from the phones (we keep a copy of the registration in the OpenSER 'location' table just for this). As far as voicemail is concerned, the OpenSER system also uses send() to send the registration to the voicemail server. I will concur with you that at this time there is no simple and quick solution to HA on *. It is what it is. I think that we are still in the womb when it comes to VoIP. Phones have become many things to many people, we have to realize that it has taken 50+ years for the phone to evolve into what it is today. Many of the features we take for granted today, (911, callerID, VoiceMail, Echo Cancelation) have only really matured in the past 15-20 years. We got a long way to go.. ___ --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] ISDN BRI and UK Premium Rate Numbers
Faris Raouf wrote: Can anyone help point me in the right direction please? I'm based in the UK and I want to start using a Premium Rate number with Asterisk - I think the equivalent in the US would be a 900 number. Effectively the caller pays much more to call such a number than a normal national or local call. The problem with these is that I don't want Asterisk to actually signal to the telephone network that the call has been answered until someone really does answer it, otherwise the caller will be paying a premium rate just to listen to an Asterisk-generated ring tone until someone answers the call. This is pretty standard Asterisk behaviour exten = whatever,1,NoOp exten = whatever,2,Dial(SIP/nSIP/n+1SIP/n+2) exten = whatever,3,Hangup The incoming ISDN call will ring the specified SIP phones, and will not be answered until one of them picks up. Snip Ideally I'd also like the caller and the person answering the call to hear a recorded message saying that calls to this number cost X per minute ... blah blah, this message being triggered only when someone answers the call. This will warn the caller *and* the person answering that this is a premium-rate call. The person answering the call will know to speak after this message has been played. But that's just an ideal situation. Right now I'm more concerned about how to stop Asterisk answering until someone is available to take the call. H ... sorry, no idea how to do this bit - I believe it's a requirement that's been addressed before by implementing a MeetMe conference, but my recollection is hazy... ___ --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] Testing IAX links
I need to test QoS on an IAX link between a server in Colorado and a server in Europe. I know I could install a Milliwatt extension on the European server and just listen, but is there a more scientific method to collect QoS metrics? Thanks P.S. I'm getting a lot of Page Not Found on lists.digium.com. Are the older posts being purged? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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] setting callerid not working if no callerid on incoming number
If we get an incoming call I can edit the callerID provided to add the leading '90' and set the name so that sales calls can be identified according to the number called. If however the callerID is unavailable then setting the callerID name or number fails (it shows as unavailable on the phone). This is the call log from such an incoming call without callerID. == Spawn extension (voip, 6204, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Zap/3-1 answered SIP/6076-30ff -- Accepting call from '' to '6201' on channel 0/2, span 1 -- Executing Set(Zap/2-1, CALLERID(number)=90) in new stack -- Executing Goto(Zap/2-1, voip|6201|1) in new stack -- Goto (voip,6201,1) -- Executing Macro(Zap/2-1, uksales|Press) in new stack -- Executing Set(Zap/2-1, CALLERID(name)=Press) in new stack -- Executing Dial(Zap/2-1, SIP/6030IAX2/6030SIP/6514|15|t) in new stack -- Called 6030 Any ideas? 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
[Asterisk-Users] Asterisk Users Group Tonight, Irvine, Ca
If you are in Southern California and would like to attend the Asterisk Users Group Meeting, it is tonight from 6-9pm at the Heritage Park Library. Irvine Heritage Park Library(949) 936-404014361 Yale AveIrvine, CA 92604 Tonight we will be having a demo of SIPX, a review of the SNOM 320 phone, and a look at FreePBX, the new version of the Asterisk Management Portal. Also, more books to give away from O'Rielly!! Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --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] Zap channel not hanging up
I see this every once in a while. I will have channels that just don't seem to hang up. When I do show channel... Elapsed Time: 24h38m15s Any suggestions? ___ --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] module load order for Junghanns qozap and TDM card
Maybe this will shed some light about what I'm trying to do: This is some output from dmesg after this load order: modprobe zaptel insmod wcfxs insmod qozap Zapata Telephony Interface Registered on major 196 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) PCI: Enabling device 02:01.0 ( - 0003) qozap: Junghanns.NET quadBRI card configured at mem 0xf889b000 IRQ 17 HZ 100 CardID 0 qozap: S/T ports: 4 [ TE TE TE TE ] qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: D-channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: D-channel (Default) (Slaves: 12) Channel 13: FXO Kewlstart (Default) (Slaves: 13) Channel 14: FXO Kewlstart (Default) (Slaves: 14) Channel 15: FXO Kewlstart (Default) (Slaves: 15) Channel 16: FXO Kewlstart (Default) (Slaves: 16) 16 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) any thoughts? Chris - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 16, 2006 10:09 AM Subject: [Asterisk-Users] module load order for Junghanns qozap and TDM card Hi all, I'm trying to get a junghanns QuadBRI to coexist in the same machine as a Digium TDM400P card (so I can run the ISDN lines in and bridge with analog phones plugged into the TDM). I'm having a problem loading the modules. If I follow the BRIstuff (0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod qozap.o I'm on Debian 2.4.31. That works. But then I still need the Digium module. (modprobe wctdm) I've tried a few different orders. Sometimes I can get the digium to load, and the qozap. but then I get an error on the ztcfg about Span invalid argument (could be my zaptel.conf I realize...) *If* I try loading the wctdm after the zaptel and qozap, the server freezes! Some loop about qozap - dropped audio card I don't know if the quadBRI and the TDM are conflicting/sharing the zaptel module, or if I need to modprobe zaptel before each of them? and in what order? Any suggestions appreciated... I haven't even got to figuring out what I can do with chan_capi, just want to get the BRI card on and stuff. Thanks for any ideas! -- Chris Earle -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Creating a voip network... use asterisk?
I wish to create a voip phone system used by different people accross the internet. I want certain people dotted around the country to be able to connect via voip to our main office. At first this will be using software phones but could extend to hardware based phones if it works well. I would like to run an asterisk server and connect everybody to this server from around the country. This could then connect to the PSTN network via ISDN in the future but initially I just want ourselves using internal calls on the voip network. Would this concept work? Is there a better way to do this? Would SIP be the protocol to use to connect remote users to our VOIP asterisk server? Would SER be a better alternative? Client | Client- The Internet---Asterisk Server --Client | | ClientPSTN Network Any help would be most appreiciated. 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
[Asterisk-Users] MeetMe - Causes * to crash :/
Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while Allison is introducing them - like playing back conf-onlyperson. This has been seen with the MeetMe participant connecting via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't seen it). The box is * 1.2.5, Zaptel 1.2.4, a TDM400P loaded with 3xFXO cards, Mandriva 2006 Free. Symptoms of the crash: once the participant hangs up, the CLI seems to freeze. One more call instance can be initiated, and the system will seize within seconds (for instance, an audio prompt will deteriorate and then stop dead). This behavior reminds me of the memory leak issue and time bomb bug, perhaps they do the same damage as this. Solution right now is to disable MeetMe, which isn't a solution as much as an amputation. Anyways, here is the CLI output, note the WARNING: alpha*CLI -- Executing Goto(SIP/Brent_ring-4473, conferences|900|1) in new stack -- Goto (conferences,900,1) -- Executing MeetMe(SIP/Brent_ring-4473, 900|sMi|1234) in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '900' -- Recording -- Playing 'vm-rec-name' (language 'en') Mar 15 16:44:38 WARNING[24014]: file.cL584 ast_readaudio_callback: Failed to write frame -- Playing 'conf-onlyperson' (language 'en') Alpha*CLI Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --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] setting callerid not working if no callerid on incoming number
Sorry forgot to mention I am running 1.2.0RC1 (dont ask :) )
Here is the macro used to set the callerid.
[macro-uksales]
; UK SALES
; ARG1 = Caller ID Name to display on phone
exten = s,1,Set(CALLERID(name)=${ARG1})
exten = s,2,Dial(SIP/6030IAX2/6030SIP/6514,15,t)
exten = s,3,Voicemail(u6030)
exten = s,4,Hangup
exten = s,103,Goto(3)
On Thu, 2006-03-16 at 16:24, Gareth Blades wrote:
If we get an incoming call I can edit the callerID provided to add the
leading '90' and set the name so that sales calls can be identified
according to the number called.
If however the callerID is unavailable then setting the callerID name or
number fails (it shows as unavailable on the phone).
This is the call log from such an incoming call without callerID.
== Spawn extension (voip, 6204, 1) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
-- Zap/3-1 answered SIP/6076-30ff
-- Accepting call from '' to '6201' on channel 0/2, span 1
-- Executing Set(Zap/2-1, CALLERID(number)=90) in new stack
-- Executing Goto(Zap/2-1, voip|6201|1) in new stack
-- Goto (voip,6201,1)
-- Executing Macro(Zap/2-1, uksales|Press) in new stack
-- Executing Set(Zap/2-1, CALLERID(name)=Press) in new stack
-- Executing Dial(Zap/2-1, SIP/6030IAX2/6030SIP/6514|15|t) in
new stack
-- Called 6030
Any ideas?
Thanks
___
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[Asterisk-Users] Dialplan : Forwarding call to voicemail after one ring iif extension is busy
Hello. Is there any way to forward incoming call to voicemail in one ring if the person on the extension is busy. Regards --- Navneet Shah Systems Administrator YL Consulting, Inc. 210-340-0098 ___ --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] Asterisk programmer needed
We are looking for an Asterisk programmer to perform maintenance and upgrading programming to a Asterisk telephony project in Indiana. You must have experience in Asterisk dialplans, digium T1 and analog hardware cards, complete knowledge of VoIP, MySQL/Asterisk integration, VoIP protocols including SIP and IAX, billing, and calling card code experience in lab testing hardware and software. Send your qualifications to [EMAIL PROTECTED] or call 574-675-7514. ___ --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] Toshiba Strata DK-280 support?
Philip Edelbrock wrote: At some point (in a few months, probably) we'll turn off the Toshiba and put viop phones on everyone's desk (including some people's at a remote office and homes). It should also cut our phone bill down to a 1/10th of what it is now! Interesting... so, you consider Asterisk / VoIP secure enough at its current stage? I have heard a lot of horror stories, and as well, I have actually experienced firsthand how bad the quality can be (I have Vonage at home, and I have had conversations from our phone system in our office with people who had VoIP systems, and the quality was pretty bad (sounded like they were underwater). This is definitely something that interests me, but I'd also be very interested in hearing others experiences with VoIP - anyone? -- Best regards, Charles ___ --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] [EMAIL PROTECTED] V's Asterisk
Hi Does anyone know the clear advantages over using asterisk rather than [EMAIL PROTECTED] Is the home version limited in anyway etc? Many thanks in Advance Scott ___ --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] [EMAIL PROTECTED] V's Asterisk
Same Asterisk but AAH is easier to setup and get running. There are no limitations. Test it out. On 3/16/06, scott [EMAIL PROTECTED] wrote: Hi Does anyone know the clear advantages over using asterisk rather than [EMAIL PROTECTED] Is the home version limited in anyway etc? Many thanks in Advance Scott ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)
On Mar 16, 2006, at 3:24 AM, Aisling wrote: x-tad-smallerHi everyone,/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerI have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don’t work. – Strange./x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAnyhow I was getting an error:/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerProcess_sdp: No compatible codecs!/x-tad-smallerx-tad-smallerAnd from the SIP debug I could see that the incoming SIP INVITE was getting a sip response of 488 Unacceptable here from my asterisk server./x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAfter doing a bit of searching I determined that this might be the fault of the codec’s particularly the G729 codec. So in the peer block that I have for my PSTN provider in my sip conf I specified allow=g729./x-tad-smallerx-tad-smallerI called my PSTN geographic number again and was delighted when the incoming calls worked. However when I next went to make an outgoing call (after having added in the “allow=g729” line), I got an infinite loop of warnings:/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerWARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)/x-tad-smallerx-tad-smallerWARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isn’t a multiple of 33 or 65 bytes long from RTP/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAfter those warnings I thought there might be a problem with the gsm codec so I commented the lines containing “allow=gsm” and still kept the line “allow=g729” because as I’ve said already incoming calls won’t work otherwise (but outgoing will)./x-tad-smallerx-tad-smallerThis however just gave another warning:/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerWARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 4 while native formats is 256 (read/write=64/64)./x-tad-smallerx-tad-smallerWhen I comment this line out again I am back to my original situation where outgoing calls work and incoming don’t./x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerHas anyone any idea how I can work around this?/x-tad-smallerx-tad-smaller /x-tad-smallerI think telling us which type of gateway is between asterisk and the PSTN might be helpful in this case... Marty ___ --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] Asterisk RealTime Question, Please help
Hi All, I will again tell what I am trying to do. I have around 1000 DID's and I have to setup context for each of it's extension and I want to do that dynamically and I do not want to change extensions.conf all the time manually whenever I want to add new context instead I will do it in Mysql DB but without mentioning that in extensions.conf asterisk is not taking it. Asterisk complains that extension is invalid as it is unable to find context in Mysql DB. It works only when I give context name and under it switch to real time. Please tell me what can I do for this? I have been searching and trying but could not get near to that :( Thanks, Manoj. Quoting [EMAIL PROTECTED]: Hi All, Thanks for your replies. I need many contexts because I have around 1000 DID's each with 5-10 Extensions. These DID numbers are changed or added very frequently and whenever there is a change I have to change Extensions.conf manually. So please tell me how can I do this dynamically without changing Extensions.conf and help me configure Asterisk. Thanks once again for your help and time, Manoj. Quoting Benchev [EMAIL PROTECTED]: I was able to install Asterisk and Asterisk-addons and use them successfully. But I have a problem now, I have many contexts and it looks like Asterisk is unable to find the context given directly in Mysql DB unless I specify it in Extensions.conf to switch it to RealTime. If I add a new context in Mysql then I have to add it in Extensions.conf and reload extensions whenever I need a new context. Please tell me if there is a way to avoid all this and make Asterisk take contexts directly from Mysql without mentioning that context in Extensions.conf. If this is possible then I can make my Asterisk RealTime actually and modify contexts directly in Mysql. Idea from the wiki: ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given peer who registers or unregisters with ; us. The actual extension is the 'regexten' parameter of the registering ; peer or its name if 'regexten' is not provided. More than one regexten may ; be supplied if they are separated by ''. Patterns may be used in regexten. ; ;regcontext=sipregistrations That means that you should creat a mother context in extensions.conf: [sipregistrations] But first I would try to add a field regcontext along with regexten(which already there) in sip_users table since for the trick to work ... read http://www.voip-info.org/wiki-Asterisk+sip+regcontext OK, that will enable the auto generation of a context but as the new context won't have a switch statement it doesn't help with this problem... I may try writing a default switch if no matching context found type patch. Well, it wont generate a context, it would rather register the extension of the new user under [sipregistrations] And, maybe now is the time to warn that regexten was created to facilitate a sip-user extensions' propagation within an * network; there is a discussion Clustering going on the list, see for details. As for the switch, since context is optional: (switch = Realtime/@realtime_ext) and if left off, RealTime will use the current context, in this case sipregistrations. Means: [sipregistrations] switch = Realtime/@realtime_ext ;realtime_ext or whatever the table name is Ok i'am guessing sans voir here since I don't understand why so many contexts are needed? Hope it helps, Benchev ___ --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] [EMAIL PROTECTED] V's Asterisk
scott wrote: Hi Does anyone know the clear advantages over using asterisk rather than [EMAIL PROTECTED] Is the home version limited in anyway etc? Using Asterisk instead of AAH gives you a better understanding of how things work and what to do when problems arise. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Dialplan : Forwarding call to voicemail after onering iif extension is busy
Sure, just make your voicemail wait 5 seconds before answering the call. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Navneet ShahSent: Thursday, March 16, 2006 10:45 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Dialplan : Forwarding call to voicemail after onering iif extension is busy Hello. Is there any way to forward incoming call to voicemail in one ring if the person on the extension is busy. Regards --- Navneet Shah Systems Administrator YL Consulting, Inc. 210-340-0098 ___ --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] module load order for Junghanns qozap and TDMcard
Okay think I finally figured this out it's the modules.conf post-install lines that run ztcfg You're not supposed to run ztcfg more than once with the multiple zaptel cards in there I kept running it manually (ztcfg -) not realizing that after modprobe wcfxs the ztcfg was being run. So the order that works is zaptel qozap wcfxs (which runs ztcfg, and readies asterisk to run) If anyone has any comments about this, please post -- Chris - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 16, 2006 11:30 AM Subject: Re: [Asterisk-Users] module load order for Junghanns qozap and TDMcard Maybe this will shed some light about what I'm trying to do: This is some output from dmesg after this load order: modprobe zaptel insmod wcfxs insmod qozap Zapata Telephony Interface Registered on major 196 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) PCI: Enabling device 02:01.0 ( - 0003) qozap: Junghanns.NET quadBRI card configured at mem 0xf889b000 IRQ 17 HZ 100 CardID 0 qozap: S/T ports: 4 [ TE TE TE TE ] qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: D-channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: D-channel (Default) (Slaves: 12) Channel 13: FXO Kewlstart (Default) (Slaves: 13) Channel 14: FXO Kewlstart (Default) (Slaves: 14) Channel 15: FXO Kewlstart (Default) (Slaves: 15) Channel 16: FXO Kewlstart (Default) (Slaves: 16) 16 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) any thoughts? Chris - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 16, 2006 10:09 AM Subject: [Asterisk-Users] module load order for Junghanns qozap and TDM card Hi all, I'm trying to get a junghanns QuadBRI to coexist in the same machine as a Digium TDM400P card (so I can run the ISDN lines in and bridge with analog phones plugged into the TDM). I'm having a problem loading the modules. If I follow the BRIstuff (0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod qozap.o I'm on Debian 2.4.31. That works. But then I still need the Digium module. (modprobe wctdm) I've tried a few different orders. Sometimes I can get the digium to load, and the qozap. but then I get an error on the ztcfg about Span invalid argument (could be my zaptel.conf I realize...) *If* I try loading the wctdm after the zaptel and qozap, the server freezes! Some loop about qozap - dropped audio card I don't know if the quadBRI and the TDM are conflicting/sharing the zaptel module, or if I need to modprobe zaptel before each of them? and in what order? Any suggestions appreciated... I haven't even got to figuring out what I can do with chan_capi, just want to get the BRI card on and stuff. Thanks for any ideas! -- Chris Earle -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] capiHOLD missing in BRIstuff 0.3.0
Hi, I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more recent version, but I cannot find any working combination of Asterisk an chan_capi any more: On junghanns.net there is a chan_capi 0.3.6, but this won't compile against any recent Asterisk (missing channel_pvt.h). The production version of BRIstuff comes with an old asterisk (1.0), the experimental version 0.3.0-PRE-1 includes an asterisk 1.2.4 and compiles, but the module capiHOLD is missing. Any ideas ? Regards, Jens ___ --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] Re: Asterisk Native Sounds - in case you missed it...
Thanks for the reference to http://winscp.sf.net/ . I always thought that it was command line, so I have always either used wget or ftp as well. I have another Linux box that I use for monitoring (mrtg and nagios) and a helpdesk system (orts) that I loaded samba on to do quick file edits, but I know that samba is a resource hog. (I would never put it on an asterisk box) I tried out winscp and love it. The link to PUTTY is a nice feature as well. And it was very easy for me to configure my favorite text editor (notepad++) . -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Bob McDowell [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] This was non-trivial for me also. I prefer to right-click-copy the link on the website, switch over to putty type in my wget (right-click), and download the file directly to the box. The link I tried on the sounds page happily downloaded index.html (if memory serves). I did go ahead and get the ulaw files the hard way... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, March 15, 2006 2:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Native Sounds - in case you missed it... On Tue, Mar 14, 2006 at 11:26:14PM -0800, Ira wrote: At 08:51 PM 03/14/2006, you wrote: In my humble opinion, EVERYONE (unless you have your own in a different voice/language) that uses Asterisk should be using these prompts. How about a direct link this time: For what it's worth, the hardest problem I had was not being able to directly FTP them from my Asterisk box. I'm a Linux newbie and had no idea how to do that. FTP *to a linux box*? /me is shocked! You have ssh access, right? Use scp/sftp. Try http://winscp.sf.net/ . If you don't one to carry one around or install on your system(s), put one statically-linked copy on your file/web server and download/run it. I downloaded them to my Windows box, set up vsftpd and uploaded them using a GUI FTP client in Windows and only then could I use them. wget http://server.name/path/to/file wget ftp://server.name/path/to/file In fact, what I normally do is copy a link from my browser to the command line in the terminal window and download it with wget. Saves me an extra file copy around the net. So for those who need exact commands, here's a two-liner: wget http://mirror.astlinux.org/sounds/asterisk-native-sounds-20060209-01-sln .tar.bz2 tar xjf asterisk-native-sounds-20060209-01-sln.tar.bz2 -C /var/lib/asterisk ___ --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
[Asterisk-Users] Re: Outbound paging dialplan example?
If you are using Comedian mail, you can to notification at the mailbox. ref: voicemail.conf 5600 = ,Steven ,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=no|saycid=no|envelope=yes|delete=no|nextaftercmd=yes The [EMAIL PROTECTED] will send an email to my phone to let me know there is a voicemail. I havent looked into Comedian mail to see if it has an urgent message option. I hope this may be helpful. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Patrick Friedel [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and doing anything with it. Right now the pertinent DP features look like this: exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,2 exten = s,5,Background(greeting) exten = 1,1,Voicemail(u100) ; Press 1 to leave a message. exten = 2,1,Voicemail(u6003) ; Press 2 to send an emergency page exten = t,1,Dial(SIP/person,30,t) ; Ring my extension on timeout Obviously extension 2 needs to be changed, right now it just leaves a message in my mailbox. I'm figuring I'll add a new message that says Please enter your callback number, followed by the pound sign. and put that in as a Background() message. The tricky bit that I can't figure out (without sample dialplans in voip-info) is how to capture the DTMF the caller provides and send it out via a System() call to an external application to page the oncall person. As the oncall person will conceivably change on a regular basis, we can't just hand it out to customers, unfortunately/thankfully. Thanks for any assistance! ___ --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] Feedback from VON expo! Info on * HA andPolycomphone!!
www.aheeva.com - Original Message - From: Jim Houser [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 16, 2006 6:50 AM Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!! Gabe. Who was the call-center program from? Thanks, Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Afana Sent: Thursday, March 16, 2006 2:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!! Hey group, I just got back from the VON expo. It was insanethere were so many companies there. The #1 thing ***EVERY*** company focused on was convergance - getting all your communication devices to intergrate with eachother. There were some nifty products out there that did some cool stuff :-) Of course Digium/Asterisk was there and I had a list of questions for them. I went by several times asking more and more questions...by the last visit, these guys were running from me because I was driving them nuts :-) Here are all the questions I asked them (this is not word for word...just a summary): Q: What are the plans for HA? A: With a configuration using DNS-SRV and DUNDi, you can create a pretty resiliant setup now. Q: What about failover without losing a call A: IBM has been able to make asterisk do this. However, at this time we are not working on any solution to offer this as part of the program. Q: Do you plan on offering support for other distros for Asterisk Business Edition? A: [uncertain answer] Not really sure...maybe SuSE...not sure Q: When is asterisk going to fully support video? A: Asterisk can complety support video using H.261, H.263 and we recently added support for H.264 Q: What do you recommend as the best solution for HA? I got two different answers for this from two different people there. Both made good sense and are basically what everyone is doing now. Here both approaces are in a nut-shell: Approach 1 (seemed to be the preferred method): Use DNS-SRV lookups for all registrations. This will distribute the calls among the * servers. Next, you configure your servers using regexten and DUNDi. You use regexten to dynamically create the exten = 1234,1,NoOp when a phone registers with that server. Then when a call comes in, you use DUNDi to try to complete the call locally. If the phone is not registered to that server, then do a DUNDi lookup to find the server that the phone is registered to and then pass the call over IAX to that server to take it to the phone. Of course the phones will need to have a short registration expiration so they update frequently because if the server they are registered to crashes, until it re-registered, no server can access it. But by doing this, the phone will re-register to another server and then the next DUNDi lookup will then go to this new server. I asked about the load of having many phones registering frequently and he said it is no big deal at all. He also said it was very important to make sure cache is disabled in DUNDi!!! Each call that is made should result in a new query. This will ensure the calls are not getting old cached info which may no longer be accurate. Approach 2: Use a SER box to handle all registrations. The SER box will take care of distributing the load between the * boxes. You do not use DUNDi or regexten in this case. Just let each * box function on its own. If one of the servers fails, SER will not use it to terminate calls. Sinces the phones are registering to SER, and all incoming calls will be routed to SER, you do not need to worry much about the * boxes. You just need to make sure you have your SER boxes in a cluster to fail-over in the event of failure. Overall theme of the Asterisk stand: selling third-party products. In the there section, Digium had 10 seperate vendors that have teamed with them to sell special programs/products/services that intergrate with Asterisk. One was a call-center program, another was a resellers package, another delt with firewalls and NAT, another for voice recognition, another was Intel (that has partnered with Digium to offer drivers in the ABE for the intel cards), another was some email, fax, chat, presence, etc. kind of box that sits in front of * to combine all these servicesand some others I dont remember. It felt like I was walking into an infomercial! I also spoke with Polycom guys a great deal and asked many questions: Q: Do you plan on offering 10/100/1000 ports on the phones? A: Yes, in the near future Q: Do you plan on offering a standard phone jack for failover purposes? A: No, we have no talks of this. However, I will take this idea to the production development team. Q: What is the services button ever used
Re: [Asterisk-Users] [EMAIL PROTECTED] V's Asterisk
At 03:04 AM 03/16/2006, you wrote: Does anyone know the clear advantages over using asterisk rather than [EMAIL PROTECTED] Is the home version limited in anyway etc? If AAH works, it's pretty cool. Personally I needed to do something it couldn't do so I gave it up after a couple of weeks. I could not see how it could handle multiple companies in one box. Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.385 / Virus Database: 268.2.4/282 - Release Date: 03/15/2006 ___ --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] Asterisk RealTime Question, Please help
I need many contexts because I have around 1000 DID's each with 5-10
Extensions.
These DID numbers are changed or added very frequently and whenever there
is a change I have to change Extensions.conf manually. So please tell me
how can I do this dynamically without changing Extensions.conf and help me
configure Asterisk.
I presume you have about 1000 DID numbers and each of this numbers may ring to
5-10 users of yours, right?
If so, make a context in you extensions.conf and include in it a switch
like that:
[ever_changing_dids]
switch = Realtime/[EMAIL PROTECTED]
Now you can insert in your extensions_table imaginary DID 9876543210:
INSERT INTO `extensions_table` VALUES ('', 'ever_changing_dids', '9876543210',
1, 'Dial', 'SIP/user1:SIP/user2:SIP/user3:SIP/user4:SIP/user8:SIP/user12|
20');
You can do that for many thousands of DIDs without changing extensions.conf.
Another approach, also no changing the extension.conf:
[ever_changing_dids]
#include includes/ever_changing_dids.conf
ever_changing_dids.conf
exten =
9876543210,1,Dial(SIP/user1SIP/user2SIP/user3SIP/user4SIP/user8SIP/user12|
20)
etc...
However this requires *CLI reload
Hope I've guessed right.
Benchev
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Re: [Asterisk-Users] Feedback from VON expo! Info on*HAandPolycomphone!!
Q: What are the plans for HA? That's BS. Last time I checked, Asterisk's support of SRV was to only grab the first SRV entry. Period. If it doesn't try any more SRV hosts after the first fails, just exactly how is that redundant? This is for the phones to fail over NOT Asterisk, remember in this case Asterisk has died so no matter what order it 'resolves' it doesn't mater in this case. I disagree. Our Asterisk boxes talk to a proxy server in certain situations. If those proxy servers where in a domain as SRV records, and one of them failed, Asterisk should try each of them in an order defined by the priority and weight. Yes, but like Alexander said, this scenerio was for the polycom to do the SRV lookup, not *. For me, the only time I will need * to do a lookup is when to hand a call off to a carrier for termination. Q: Whats the best way to program the phone to handle failover? A: Use a DNS-SRV address for the primary server. When the phone queries the DNS server, it will receive a list of all the possible servers This is broken to some degree. When the phone refreshes it's cache, and grabs the list of SRV servers again, it will continue to use them in the same manner until it refreshes it's cache again, or there is a failure, even when all SRV hosts have the same priority and weight. It should round robin in this case. Agreed. This is how the polycom guy explain it. Lets say you do an srv lookup and get: sip1.test.com sip2.test.com sip3.test.com sip4.test.com The phone will try to register with sip1.test.com. If it is successful, great. If not, continue to sip2.test.com, then sip3, sip4 and then back again to sip1 and it will cycle untile it can find a server to register with. Now lets say you are registered to sip1.test.com, if you pick up the phone to make a call, it will try to send it to sip1.test.com. If the call fails to go through, the phone will then try to send the call through sip2, then sip3, sip4..until it can make the call (just like for registration). This will not cause it to re-register however. It will not register until its registration expires and it has to re-register. At this time it will refer back to the same SRV lookup and continue through the list. I just thought now that this could cause issues because if all phones get the SRV lookup saying sip1, sip2, sip3 and sip4 in that order, all phones will register to sip1 if they can. If the priority and weight is set the same, will the SRV lookup return these servers in a round-robin or even random way? And in regards to Asterisk HA, and approach #2. If you have your SER boxes use the send() command to stateless forward registrations, you can send registrations from the phones to ALL your Asterisk systems so that every Asterisk box knows about every phone, and every Asterisk box can route calls from/to any phone. Then you have issues with hints, voicemail, and other features. Hints, voicemail and other features, to this point, are all working fine. The OpenSER systems routes SUBSCRIBE/NOTIFY/MESSAGE etc messages to /from the phones (we keep a copy of the registration in the OpenSER 'location' table just for this). As far as voicemail is concerned, the OpenSER system also uses send() to send the registration to the voicemail server. I would rather stay away from SER if I can because its complicated to get setup (no big deal though), but it ads another layer to the process and creates a single point of failure. You can have a few SER machines in a linux cluster to fail-over, but this can take up to several seconds and is unacceptable since doing this time, *no* calls can go in or out. At least with the DNS model, you know a DNS lookup will work (just have a primary, secondary...etc - something will work) and if a server fails, it doesn't criple the whole service. - Gabe ___ --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] Action after _caller_ has hungup(cmd Dial 'g'-option)
On Mon, 13 Mar 2006 14:19:05 +0200 Benchev [EMAIL PROTECTED] wrote: Hmm, both of you recommend a solution with the dial cmd in an agi-script, i would prefer a direct solution but i guess there is none. There is - H - Allow the calling party to hang up by hitting the '*' DTMF digit. I though that your main concern was how to cachup the hangup and deal with the result of a call(see my previous email ), which is bigger pain than H. Sorry misunderstanding you. Benchev no no, you've understood perfectly right. i want to handle the call after the caller has hungup, just as deadagi does. in the meanwhile i've tried to write a deadagi for this, but since my dialstring includes pipes(Zap/G1/123423|120|gA(xyz)) it's a real pain in the ass to send this to asterisk(asterisk swallows the | and misinterprets \|) a direct solution, like the g option would be great, but i guess i'm out of luck here... :-( thanks for your help though... regards chris ___ --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] capiHOLD missing in BRIstuff 0.3.0
[EMAIL PROTECTED] wrote: Hi, I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more recent version, but I cannot find any working combination of Asterisk an chan_capi any more: On junghanns.net there is a chan_capi 0.3.6, but this won't compile against any recent Asterisk (missing channel_pvt.h). The production version of BRIstuff comes with an old asterisk (1.0), the experimental version 0.3.0-PRE-1 includes an asterisk 1.2.4 and compiles, but the module capiHOLD is missing. Did you try searching for the chan capi within the mailing list archive before posting? http://sourceforge.net/projects/chan-capi is the current way to go with a CAPI capable card. hth -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] capiHOLD missing in BRIstuff 0.3.0
On Thu, 16 Mar 2006, Peer Oliver Schmidt wrote: [EMAIL PROTECTED] wrote: Hi, I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more recent version, but I cannot find any working combination of Asterisk an chan_capi any more: On junghanns.net there is a chan_capi 0.3.6, but this won't compile against any recent Asterisk (missing channel_pvt.h). The production version of BRIstuff comes with an old asterisk (1.0), the experimental version 0.3.0-PRE-1 includes an asterisk 1.2.4 and compiles, but the module capiHOLD is missing. Did you try searching for the chan capi within the mailing list archive before posting? http://sourceforge.net/projects/chan-capi is the current way to go with a CAPI capable card. chan-capi.org will be online soon. sourceforge will then be obsolete. The chan-capi packages are already available on ftp://ftp.chan-capi.org Armin ___ --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] Budgetone strange problem - have to press hold on and off to connect call.
I have a strange problem in that I have put a budgetone out on the internet that connects to my * server that's behind a firewall. They can call me I can call them and it works fine. However, I have setup a link to sipdiscount on my * server. If the budgetone user calls via my * box to sipdiscount all the budgetone user hears is silence and the called person hears silence as well when they pick up the phone. If the budgetone user then hits the hold button then the called party hears the music on hold and then when the budgetone users takes it off hold then they can start talking to each other! The phone has firmware 1.08.16 same problem with reinvite yes or no. Any ideas? Chris ___ --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] Re: transfers/parked calls + polycom 501
Hi - I am not sure what I did but blind transfers do not work. The Polycom does not allow me to dial the extension of the person I want to transfer to after I hit: transfer - blind I would strongly suggest getting the latest firmware, and using the sample configuration files with that firmware to set up your phone. This SHOULD work. If it still does not work after doing this, there may be a hardware issue with your phone. - Noah ___ --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] Re: transfers/parked calls + polycom 501
I am using the latest firmware and bootrom and this is a problem with all 12 polycom 501s that we have in the office. If I want to transfer to 1005 for example while on the phone with the original caller, I press transfer - blind - type "1", "0" then the phone clears the display and the transfer fails. It only allows me to dial the first two digits of the extension I want to transfer to. It even happens when I dial local sip to local sip, not just sip to pstn. This seems like a config mistake I made.thanks Noah Miller [EMAIL PROTECTED] wrote: Hi - I am not sure what I did but blind transfers do not work. The Polycom does not allow me to dial the extension of the person I want to transfer to after I hit: transfer - blindI would strongly suggest getting the latest firmware, and using the sampleconfiguration files with that firmware to set up your phone. This SHOULDwork. If it still does not work after doing this, there may be a hardwareissue with your phone.- Noah___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Brings words and photos together (easily) with PhotoMail - it's free and works with Yahoo! Mail.___ --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
