Re: [asterisk-users] Detect if a Number is up or not
another idea you could test is to use a very short Timeout in your Dial command. like Dial(ZAP/012345678,1) - will dial and exit after 1 sec with DIALSTATUS set accordingly HTH, Ioan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing For AsteirskAddon
when i install asterisk addon ,i got error here chan_ooh323.c:1934: error: dereferencing pointer to incomplete type chan_ooh323.c:1935: error: dereferencing pointer to incomplete type chan_ooh323.c:1937: error: dereferencing pointer to incomplete type chan_ooh323.c:1938: error: dereferencing pointer to incomplete type chan_ooh323.c:1940: error: dereferencing pointer to incomplete type chan_ooh323.c:1943: error: dereferencing pointer to incomplete type Thanks for your help -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Message notification without MWI
Hello, For a couple of hardphones which do not have Message Waiting Indicator, I'm wondering what could be the most efficient and reliable way to notify users a message is waiting. Though messages could be sent as email attachment, I'm thinking I should mimic cellphones behaviour like this: 5 min after a message if left, then call back mailbox owner and let him listen to its message, 30 min after message drop, another call 4 hours after message drop, another call 24 hours after message drop, the last notification call occurs, if a notification call succeeds, then every other planned notification call is removed, if user checks its voicemail, then every planned notification call is removed. What do you think of this ? To generate notification calls, call files and externnotify parameter in voicemail.conf seems to fit. What do you think of using it ? What about inotify tools ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit
Steve Gladden wrote: So that explains why it won't compile eh? And wow Kevin... I'm curious how much work would it be and would it be worth it? I've always imagined RT kernels would be excellent for asterisk. I've also wondered why it appears not to have been done 'out there' Or discussed very much. It's never possible to state why something has *not* been done, unless there is a very clear reason for not doing it. I don't know that anyone can state how much work it would be, since the RT kernels have been a wildly moving target for most of the past two years. Whether the work would be worth the effort or not cannot be known until the amount of work is known, so... As far as usage in the Asterisk community, the vast majority of Asterisk servers run Asterisk itself as the single primary application on the system, so there really isn't competition for CPU resources between Asterisk and other applications (which is where RT kernels would possibly be helpful). Whenever someone reports issues because they have a busy database server, web server, or some other application on the same box, the conventional approach is to get that other application onto another box. It is conceivable that RT kernels will help to eliminate the need to do that, but until they have reached mainline status, it's doubtful anyone will do the work to support them (not that Asterisk needs any work in this area anyway... it's DAHDI, which is not necessary in all Asterisk installations). Finally... it seems highly unlikely that an RT kernel would be able to provide any tangible benefits when running in a virtualized system; the entire point of RT kernels is to be able to guarantee CPU availability on a predictable schedule, which isn't possible when the virtualization system can't guarantee that itself. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan question
Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong? extensions.conf: [office] exten = 601,1,Answer() exten = 601,2,Wait,2 exten = 601,3,Dial(SIP/601,20) exten = 601,4,Hangup() exten = 500,1,Answer() exten = 500,2,Wait,2 exten = 500,3,Dial(SIP/500,20) exten = 500,4,Hangup() sip.conf: [601] deny=0.0.0.0/0.0.0.0 context=office type=friend secret=601 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/601 canreinvite=no callgroup=1 pickupgroup=1 callerid=device 601 accountcode= call-limit=50 [500] deny=0.0.0.0/0.0.0.0 username=500 context=office type=friend secret=500 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/500 canreinvite=no callgroup=1 pickupgroup=1 callerid=device 500 accountcode= call-limit=50 -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect 2 asterisks servers
Hi! I need some help Well i have this cenario: 1 ip04 running asterisk [A] 1 pc running asterisk [B] I nedd to make calls from A to B, and B to A. Via sip The A-B calls are working. Now I need to configure the dial plan to call B-A either to sip numbers and Fxs. Anyone can help me? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect 2 asterisks servers
1. Why aren't you using IAX instead of SIP/FXS? 2. If you can connect from A-B using SIP, the process should be reversible unless B just sees A as a phone and not a peer/server. 3. to make an FXS connection, you're going to have to introduce Zaptel/DAHDI (you don't state what level of * you are on) to the equation. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of matheus coppetti Sent: Tuesday, April 27, 2010 7:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Connect 2 asterisks servers Hi! I need some help Well i have this cenario: 1 ip04 running asterisk [A] 1 pc running asterisk [B] I nedd to make calls from A to B, and B to A. Via sip The A-B calls are working. Now I need to configure the dial plan to call B-A either to sip numbers and Fxs. Anyone can help me? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect if a Number is up or not
This is probably a good idea, BUT it is likely that the dialed phone will never ring (Perhaps that is the desired effect); In my experience it takes Zap/DAHDI about 2-7 seconds to generate the first ring of a call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias Sent: Tuesday, April 27, 2010 2:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Detect if a Number is up or not another idea you could test is to use a very short Timeout in your Dial command. like Dial(ZAP/012345678,1) - will dial and exit after 1 sec with DIALSTATUS set accordingly HTH, Ioan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan question
In your sip.conf your permit line does not have an ip address to allow the register from so the call is coming in as a guest and that is likely using context default. Set the permit line to either the ip address of the phone or the network the phone is on. permit=192.168.1.0/255.255.255.0 with allow from any 192.168.1.x address as an example. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 4:31 AM, Vasiliy G Tolstov wrote: Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong? extensions.conf: [office] exten = 601,1,Answer() exten = 601,2,Wait,2 exten = 601,3,Dial(SIP/601,20) exten = 601,4,Hangup() exten = 500,1,Answer() exten = 500,2,Wait,2 exten = 500,3,Dial(SIP/500,20) exten = 500,4,Hangup() sip.conf: [601] deny=0.0.0.0/0.0.0.0 context=office type=friend secret=601 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/601 canreinvite=no callgroup=1 pickupgroup=1 callerid=device 601 accountcode= call-limit=50 [500] deny=0.0.0.0/0.0.0.0 username=500 context=office type=friend secret=500 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/500 canreinvite=no callgroup=1 pickupgroup=1 callerid=device 500 accountcode= call-limit=50 -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound route question
Simply place the SIP Extension of the GSM gateway in another context context=from-gsm and in your extensions.conf use something like this [from-gsm] exten= = _X.,1,Goto(whatever IVR you want) Date: Mon, 26 Apr 2010 17:23:40 -0300 From: aco1...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Inbound route question Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and 1002) and a GSM Gateway with SIP extension . Two cell phones call to the GSM Gateway number and after that they get a ring tone to dial to the SIP extensions. Is it possible to consider the GSM Gateway SIP extension as an incoming call to the Asterisk PBX and so create an inbound route that point: GSM Gateway DID: - IVR in order to point all incoming cell phone calls to my existing IVR ??? Thanks a lot. Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan question
В Втр, 27/04/2010 в 06:11 -0700, Jim Dickenson пишет: In your sip.conf your permit line does not have an ip address to allow the register from so the call is coming in as a guest and that is likely using context default. Set the permit line to either the ip address of the phone or the network the phone is on. permit=192.168.1.0/255.255.255.0 with allow from any 192.168.1.x address as an example. Thank You. But a get work with this lines: exten = 102,1,Answer() exten = 102,2,Dial(SIP/102,20) exten = 102,3,Hangup() exten = 500,1,Answer() exten = 500,2,Dial(SIP/500,20) exten = 500,3,Hangup() exten = 601,1,Answer() exten = 601,2,Dial(SIP/601,20) exten = 601,3,Hangup() -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems for Skype for Asterisk
Is there an issue with running it with the latest from the 1.6.2 branch? I did an svn update and make install and now when somebody comes in via Skype, I get an infinite loop of: [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument and there's no voice path in either direction. If there's an issue, what's the latest svn revision number I can use? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems for Skype for Asterisk
Richard Kenner wrote: Is there an issue with running it with the latest from the 1.6.2 branch? I did an svn update and make install and now when somebody comes in via Skype, I get an infinite loop of: [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument and there's no voice path in either direction. If there's an issue, what's the latest svn revision number I can use? Digium's commercial add-on modules for Asterisk are only tested with numbered releases of Asterisk; when API changes occur in a branch, we try to ensure that an updated version of the add-on module is made available very close to the time that change makes its way into a numbered release. While I can't say that is the cause of the issue you are seeing, if the problem does not occur using the latest release from the 1.6.2 branch, you'll probably have to wait until the next release is made for an updated Skype For Asterisk release to go along with it. If it does occur using the latest release, then you should contact Digium Support to report the issue so it can be expedited to the Skype For Asterisk maintainers. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E3 Card on Asterisk ?
Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ? Thanks for any input you may provide. regards, Anita Hall, Simmortel Voice. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Record call without caller interference
Hello list, can a conversation be recorded without the caller or callee having to press some combination that is defined in features.conf ?? Like in queues.conf you have the ability to record a conversation with MixMonitor when the caller is connected to an agent/member of the queue. Can this auto-recording also be implied on normal Dial(something) ?? So that when the call is picked up (and the 2 parties are connected together) the recording begins... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E3 Card on Asterisk ?
Anita Hall wrote: Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ? Thanks for any input you may provide. regards, Anita Hall, Simmortel Voice. On ONE box? Seems to me, that is just asking for trouble Check the archives, others have suggested maximums for a single box and server farms John Novack Checked by AVG - www.avg.com Version: 9.0.814 / Virus Database: 271.1.1/2837 - Release Date: 04/26/10 14:27:00 -- Dog is my co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E3 Card on Asterisk ?
- Anita Hall anita.h...@simmortel.com wrote: Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ? Thanks for any input you may provide. regards, Anita Hall, Simmortel Voice. Putting that many calls on one system is not a good idea... you'll want to spread those calls over multiple machines with some sort of redundancy or failover. Also, the card you reference is for clear channel data only, not voice. However, you may want to give Sangoma a call. They are top notch when it comes to custom solutions and have some of the brightest people on staff. In my research into T3/E3 connectivity, I've found it's generally best to go with some sort of a MUX that will take your T3/E3 and break it down into individual T1/E1 circuits. If I recall, Adtran makes a few nice units for this. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E3 Card on Asterisk ?
On 04/27/2010 10:41 PM, Anita Hall wrote: Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ? Thanks for any input you may provide. That card only operates as an E3 or T3 data card. It is not channelised to work as a PSTN voice card. So, no, it doesn't work with Asterisk or Freeswitch - unless you are looking for IP connecion for VoIP. Sangoma used to talk about producing a channelised revision of the card, but it looks like the potential sales have never looked promising enough to make it happen. Digium used to have an E3/T3 card on their web site, but I assume they canned it. You don't hear anything about it now. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record call without caller interference
Jonas Kellens wrote: Hello list, can a conversation be recorded without the caller or callee having to press some combination that is defined in features.conf ?? Like in queues.conf you have the ability to record a conversation with MixMonitor when the caller is connected to an agent/member of the queue. Can this auto-recording also be implied on normal Dial(something) ?? So that when the call is picked up (and the 2 parties are connected together) the recording begins... Kind regards, Jonas. You can use MixMonitor and use it's option b which only starts recording once the call has been bridged Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record call without caller interference
Yers. You have 2.5 options: Monitor, MixMonitor, (these make 1,5) and JACK_HOOK On Tue, Apr 27, 2010 at 5:53 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello list, can a conversation be recorded without the caller or callee having to press some combination that is defined in features.conf ?? Like in queues.conf you have the ability to record a conversation with MixMonitor when the caller is connected to an agent/member of the queue. Can this auto-recording also be implied on normal Dial(something) ?? So that when the call is picked up (and the 2 parties are connected together) the recording begins... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callprogress issue
I'm running Asterisk 1.6.2.1 with DAHDI 2.2.1 using a TDM410P. I have callprogress=yes in chan_dahdi.conf because, from everything I've read, it is needed when using call files over PSTN, which I DO use occasionally. I know that callprogress=yes is experimental and causes some issues. We've never experienced any problems when making local calls over PSTN with callprogress turned on. However, 1-800 calls do not work (and possibly other long-distance calls but I do not have long distance service on my PSTN and can't test that). My Dial() command has a ring-timeout of 60 seconds. I've noticed that calls to 1-800 numbers cut off after 60 seconds even if you're in the middle of a conversation. I've also had problems getting DTMF signals to work when making 1-800 calls which makes it impossible to interact with menus during the call. None of these problems occur when the calls are within our own local calling area. Could this be a callprogress issue and why would it only cause problems for non-local calls? -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan question
I am not sure what you are asking here. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 6:16 AM, Vasiliy G Tolstov wrote: В Втр, 27/04/2010 в 06:11 -0700, Jim Dickenson пишет: In your sip.conf your permit line does not have an ip address to allow the register from so the call is coming in as a guest and that is likely using context default. Set the permit line to either the ip address of the phone or the network the phone is on. permit=192.168.1.0/255.255.255.0 with allow from any 192.168.1.x address as an example. Thank You. But a get work with this lines: exten = 102,1,Answer() exten = 102,2,Dial(SIP/102,20) exten = 102,3,Hangup() exten = 500,1,Answer() exten = 500,2,Dial(SIP/500,20) exten = 500,3,Hangup() exten = 601,1,Answer() exten = 601,2,Dial(SIP/601,20) exten = 601,3,Hangup() -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems for Skype for Asterisk
We are running Asterisk 1.6.2.7-rc1 and SfA without problem. What version are you running? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner Sent: Tuesday, April 27, 2010 9:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problems for Skype for Asterisk Is there an issue with running it with the latest from the 1.6.2 branch? I did an svn update and make install and now when somebody comes in via Skype, I get an infinite loop of: [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument and there's no voice path in either direction. If there's an issue, what's the latest svn revision number I can use? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems for Skype for Asterisk
We are running Asterisk 1.6.2.7-rc1 and SfA without problem. What version are you running? I'm using the current version from the 1.6.2 SVN branch, which is called SVN-branch-1.6.2-r258676M. I'm glad to know that 1.6.2.7-rc1 works because that's closer to what I have than 1.6.2.6. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect 2 asterisks servers
all you need to do is make the configurations mirror each other.
in the example below, all of the endpoints are SIP, but it doesn't matter if
you move the endpoints to another protocol, like Fxs:
on serverA
extesions.conf:
[phones]
include = localphones
include = to_serverB
[localphones]
exten = _11X,1,NoOp()
exten = _11X,n,Dial(SIP/${EXTEN},30)
exten = _11X,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _11X,n,Hangup()
[to_serverB]
exten = _12X,1,NoOp()
exten = _12X,n,Dial(SIP/serverB/${EXTEN})
exten = _12X,n,Hangup()
[from_serverB]
include = localphones
sip.conf:
register = serverA:secret@ip_of_serverB/serverB
[serverB]
type=friend
secret=secret
context=from_serverB
host=dynamic
[sets](!)
type=friend
context=phones
host=dynamic
[110](sets)
[111](sets)
[112](sets)
###
on serverB
extesions.conf
[phones]
include = localphones
include = to_serverA
[localphones]
exten = _11X,1,NoOp()
exten = _11X,n,Dial(SIP/${EXTEN},30)
exten = _11X,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _11X,n,Hangup()
[to_serverA]
exten = _12X,1,NoOp()
exten = _12X,n,Dial(SIP/serverA/${EXTEN})
exten = _12X,n,Hangup()
[from_serverA]
include = localphones
sip.conf:
register = serverB:secret@ip_of_serverA/serverA
[serverA]
type=friend
secret=secret
context=from_serverA
host=dynamic
[sets](!)
type=friend
context=phones
host=dynamic
[120](sets)
[121](sets)
[122](sets)
[123](sets)
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of matheus coppetti
Sent: Tue 4/27/2010 5:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Connect 2 asterisks servers
Hi!
I need some help
Well i have this cenario:
1 ip04 running asterisk [A]
1 pc running asterisk [B]
I nedd to make calls from A to B, and B to A. Via sip
The A-B calls are working. Now I need to configure the dial plan to call B-A
either to sip numbers and Fxs.
Anyone can help me?
winmail.dat--
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[asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1
All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=551234 secret=password defaultuser=551234 insecure=port,invite context=broadvoice authname=551234 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no monitor=yes qualify=yes disallow=all allow=ulaw nat=yes register = 551...@sip.broadvoice.com:password:551...@sip.broadvoice.com in extensions.conf: [broadvoice] exten = 551234,1,Set(CDR(accountcode)=44) exten = 551234,n,AppendCDRUserField(BroadVoice) exten = 551234,n,7090093,1,Goto(112,1) and Asterisk is still giving me this error in the logs (while playing a number does not exist sound clip): [Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to extension '551234' rejected because extension not found. I have played with the register settings, I have played with the sip context settings, I have tried an 's' extension in the broadvoice context, and I am out of ideas. Does anyone have an idea of what is going on with this? Thanks, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1
[sip.broadvoice.com]
...
[broadvoice]
exten = 551234,1,Set(CDR(accountcode)=44)
and Asterisk is still giving me this error in the logs (while playing a
number does not exist sound clip):
[Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to
extension '551234' rejected because extension not found.
I have played with the register settings, I have played with the sip
context settings, I have tried an 's' extension in the broadvoice
context, and I am out of ideas. Does anyone have an idea of what is
going on with this?
The register is irrelevant for incoming calls and an 's' extension
won't get reached in this situation. MOST LIKELY what's happening is
that the SIP call isn't maching the security parameters in
[sip.broadvoice.com] and thus being put into the default context.
To test this theory, add
exten = _X.,1,NoOp(${EXTEN})
in both the default and broadvoice contexts and see which one gets hit
and what the extension is when you make the incoming call.
If it's going to default, then turn SIP debugging on and then make
another call and see if the parameters in the INVITE match what you
expect in the [sip.broadvoice.com] clause.
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Re: [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1
Is this an inbound call to that number? Or are you calling out from that number? I understand the need to obfuscate the numbers, but it says Call from '551234' to extension '551234', so are you calling yourself? Or did you just change both numbers to the same number. Maybe just change the first 6 digits, so we can read it easier. And more debug info would help. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark Sent: Tuesday, April 27, 2010 7:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1 All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=551234 secret=password defaultuser=551234 insecure=port,invite context=broadvoice authname=551234 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no monitor=yes qualify=yes disallow=all allow=ulaw nat=yes register = 551...@sip.broadvoice.com:password:551...@sip.broadvoice.com in extensions.conf: [broadvoice] exten = 551234,1,Set(CDR(accountcode)=44) exten = 551234,n,AppendCDRUserField(BroadVoice) exten = 551234,n,7090093,1,Goto(112,1) and Asterisk is still giving me this error in the logs (while playing a number does not exist sound clip): [Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to extension '551234' rejected because extension not found. I have played with the register settings, I have played with the sip context settings, I have tried an 's' extension in the broadvoice context, and I am out of ideas. Does anyone have an idea of what is going on with this? Thanks, Seann -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E3 Card on Asterisk ?
On Tue, 2010-04-27 at 11:01 -0400, John Novack wrote: Anita Hall wrote: Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ? Thanks for any input you may provide. regards, Anita Hall, Simmortel Voice. On ONE box? Seems to me, that is just asking for trouble Check the archives, others have suggested maximums for a single box and server farms John Novack Has anyone put together a public list/wiki/info sheet on what the various maximums/rules of thumb are? Seems a better idea than random searching to point to a definitive document! And save some traffic to the list as this seems to be a common query. BillK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial plan question.
Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : al...@pbx.com should be able to call b...@pbx.com where pbx.com is astersik. Can you pl let me know how I can achieve this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1
The hidden number is no different from what I posted. This is inbound, I pick up my cell phone, dial 551234, which then hits my * box, which then the * box barfs that error. On 4/27/2010 8:35 PM, Peder wrote: Is this an inbound call to that number? Or are you calling out from that number? I understand the need to obfuscate the numbers, but it says Call from '551234' to extension '551234', so are you calling yourself? Or did you just change both numbers to the same number. Maybe just change the first 6 digits, so we can read it easier. And more debug info would help. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark Sent: Tuesday, April 27, 2010 7:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1 All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=551234 secret=password defaultuser=551234 insecure=port,invite context=broadvoice authname=551234 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no monitor=yes qualify=yes disallow=all allow=ulaw nat=yes register = 551...@sip.broadvoice.com:password:551...@sip.broadvoice.com in extensions.conf: [broadvoice] exten = 551234,1,Set(CDR(accountcode)=44) exten = 551234,n,AppendCDRUserField(BroadVoice) exten = 551234,n,7090093,1,Goto(112,1) and Asterisk is still giving me this error in the logs (while playing a number does not exist sound clip): [Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to extension '551234' rejected because extension not found. I have played with the register settings, I have played with the sip context settings, I have tried an 's' extension in the broadvoice context, and I am out of ideas. Does anyone have an idea of what is going on with this? Thanks, Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan question.
I am not sure what your problem is. You can have a numeric extension dial an alphabetic sip user. exten = 123,1,Dial(SIP/somename) The soft phone registers to your box with whatever username you set up. If your phone can dial alpha then you can have exten = alpha,1,Dial(SIP/$(EXTEN}) -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote: Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : al...@pbx.com should be able to call b...@pbx.com where pbx.com is astersik. Can you pl let me know how I can achieve this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broadvoice inbound fails on Asterisk 1.6.1
On 4/27/2010 8:07 PM, Richard Kenner wrote:
[sip.broadvoice.com]
...
[broadvoice]
exten = 551234,1,Set(CDR(accountcode)=44)
and Asterisk is still giving me this error in the logs (while playing a
number does not exist sound clip):
[Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to
extension '551234' rejected because extension not found.
I have played with the register settings, I have played with the sip
context settings, I have tried an 's' extension in the broadvoice
context, and I am out of ideas. Does anyone have an idea of what is
going on with this?
The register is irrelevant for incoming calls and an 's' extension
won't get reached in this situation. MOST LIKELY what's happening is
that the SIP call isn't maching the security parameters in
[sip.broadvoice.com] and thus being put into the default context.
To test this theory, add
exten = _X.,1,NoOp(${EXTEN})
in both the default and broadvoice contexts and see which one gets hit
and what the extension is when you make the incoming call.
If it's going to default, then turn SIP debugging on and then make
another call and see if the parameters in the INVITE match what you
expect in the [sip.broadvoice.com] clause.
Did that, didn't get very far with the dialplan route. Checked the
invite settings and realized, duh on me, that the domain wasn't matching
up. It was passing my public IP address, and * was looking for the
asterisk box IP. Changed that setting and tested and it works.
Thanks for the idea's and helping rattle a bit more sense into what I
was doing.
~Seann
smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] Dial plan question.
Thanks a lot jim for the reply. My issue is : there is no numbers involved. I have soft clients. when a user (bob) calls Alex, he just opens his sip client and types in a...@pbx.com so how can I write the translations for a case like that? the examples you gave are when there are numbers..can u pl give me complete numbering plam From: Jim Dickenson dicken...@cfmc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, April 27, 2010 7:09:45 PM Subject: Re: [asterisk-users] Dial plan question. I am not sure what your problem is. You can have a numeric extension dial an alphabetic sip user. exten = 123,1,Dial(SIP/somename) The soft phone registers to your box with whatever username you set up. If your phone can dial alpha then you can have exten = alpha,1,Dial(SIP/$(EXTEN}) -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote: Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : al...@pbx.com should be able to call b...@pbx.com where pbx.com is astersik. Can you pl let me know how I can achieve this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BN8S0, dahdi, wcb4xxp
Hi, a few month ago, I tried to install zaptel for my Beronet BN8S0 pci card... I gave up and took hfcmulti/lcr. Now dahdi (2.2.1.1) seems to support the card and I'm very interested to get it to work. But how to get rid of these annoying qozap driver? bishop dahdi # lspci -v -nn -s 01:00.0 01:00.0 ISDN controller [0204]: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] [1397:16b8] (rev 01) Subsystem: Cologne Chip Designs GmbH Device [1397:b562] Flags: medium devsel, IRQ 21 I/O ports at 9480 [size=8] Memory at fb9bb000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel modules: wcb4xxp, hfcmulti bishop dahdi # modprobe wcb4xxp bishop dahdi # lspci -v -nn -s 01:00.0 01:00.0 ISDN controller [0204]: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] [1397:16b8] (rev 01) Subsystem: Cologne Chip Designs GmbH Device [1397:b562] Flags: medium devsel, IRQ 21 I/O ports at 9480 [size=8] Memory at fb9bb000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel driver in use: wcb4xxp Kernel modules: wcb4xxp, hfcmulti bishop dahdi # dahdi_hardware -v driver should be 'qozap' but is actually 'wcb4xxp' pci::01:00.0 qozap+ 1397:16b8 Junghanns OctoBRI ISDN card Completely confused: Clairet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan question.
here is the dail plan I am using:
my extensions file:
[globals]
[ext-sip]
host=provider.sip.com
[default]
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
expected dialing plan:
when some one calls bob,
Asterisk should add b...@provider.sip.com and sent to the external world.
But that is not working,.
can you pl let me know what I am missing?
Also,
is there a way that Asterisk will read completely b...@provider.sip.com from
the received sip message and forwards directly to that domain.
That means,
When we receive a Request to b...@provider.sip.com, Asterisk should send that
to the outgoing interface to b...@provider.sip.com\.
some plan like..
extern=b...@x.com,1,Dial(SIP/{EXTERN},20)...
From: Aditya Kumar adityakumar...@yahoo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, April 27, 2010 10:11:16 PM
Subject: Re: [asterisk-users] Dial plan question.
Thanks a lot jim for the reply.
My issue is :
there is no numbers involved. I have soft clients.
when a user (bob) calls Alex,
he just opens his sip client and types in a...@pbx.com
so how can I write the translations for a case like that?
the examples you gave are when there are numbers..can u pl give me complete
numbering plam
From: Jim Dickenson dicken...@cfmc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, April 27, 2010 7:09:45 PM
Subject: Re: [asterisk-users] Dial plan question.
I am not sure what your problem is. You can have a numeric extension dial an
alphabetic sip user.
exten = 123,1,Dial(SIP/somename)
The soft phone registers to your box with whatever username you set up.
If your phone can dial alpha then you can have
exten = alpha,1,Dial(SIP/$(EXTEN})
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote:
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved .
so How can I make the configuration to work ( with numbers I can get this
done using extensions.conf)
my expected result is :
al...@pbx.com should be able to call b...@pbx.com
where pbx.com is astersik.
Can you pl let me know how I can achieve this?
--
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