Re: [asterisk-users] Change Packetization Time
- Original Message - From: Dan Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 19, 2007 7:58 PM Subject: Re: [asterisk-users] Change Packetization Time Dovid wrote: Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. Your client is correct. Configurable packetization was added introduced with the release of 1.4.0. For details look at the rtp-packetization.txt file in the doc directory for full details. The short answer is to append :size to any codec on your allow directive that you want to change from the default of 20ms. Ex. Allow=g729:40 Dan Dan, Can I make this change in 1.2.X ? (maybe in the source ?). I have not moved to 1.4.X because of the lack of support. Currently using SpanDSP. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
On Sun, Aug 19, 2007 at 09:00:33PM -0400, Matthew Brothers wrote: Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? I wouldn't exactly say that it is too difficult but that the target audience for the default examples is not the average person/entity that could make use of the power inherent with DUNDi. When an average * user/admin wants to use DUNDi they will want to start out small and local rather than worry about all of the intricacies of the e164 standard. It is much easier, in my opinion, to learn the power of DUNDi on a simple level and scale that up to a more globally connected platform. I'd say that duni.conf is a reference, and you expect it to be an introductory document. A reference should be comprehensive. It is best used after you've grasped the basic concepts, and together with a text search. Asterisk's sample configuration files actually serve a role of a reference. If you were to look for an introduction-level document in the asterisk source, you should have started in the /doc directory. Sadly the documentation there is close to non-existing at the moment: http://www.asterisk.org/doxygen/1.4/AstDUNDi.html How did I find that page? I went to the doxygen-generated documentation for 1.4: http://www.asterisk.org/doxygen/1.4/ In there, one non-trivial jump to the rest of the interesting documentation: Related Pages And there I can find some pretty handy documentation. If you have anything more to comment on that, I guess the place for that is either the (practically dead) asterisk-doc mailing list, or looking at some of the work done on the admin guide for 1.6 . (yeah, I know, patches are welcome, docs talk, whatever) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 asterisk servers, how to connect them together?
You may want to start from here: http://astrecipes.net/index.php?n=204 l. On Sun, 19 Aug 2007 00:46:45 +0200, Ade Vickers [EMAIL PROTECTED] wrote: Hi... I have what is, I am sure, a relatively common straightforward problem (no, NOT that kind of problem!)... I'm trying to hook two asterisk servers together so I can make a distributed PBX. Here's the scenario: [MASTER] is in the office. It has unrestricted access to the internet, and a fixed IP address. It has 3 SIP hardphones configured working, plus a couple of softphones which log in/out as necessary. The phones are on extensions 5100-5104, with a special extension 5999 which just plays music. [HOME] is at home. It has internet access only through a Microsoft ISA 2003 firewall, and has a dynamic IP address. It has 1 SIP hardphone configured, and working, on extension 5110. I can add a second hardphone to verify that this (new build) server is working OK, but all of the messages indicate it's fine. What I want to do, obviously, is have ALL of the extensions (5XXX) pretending to be on the same PBX. i.e. if I dial 5100 (on [MASTER]) from 5110 (on [HOME]), the call goes through everyone's happy; and vice versa, calling 5110 from 5100. I know I need to use IAX to achieve this (as IAX can negotiate its way past the firewall), but I can't find the magic incantations for IAX.CONF (on either server) to make them talk nicely to each other. They did, very briefly, as the [MASTER] server spotted the IP address of [HOME], added it to the peer list, my heart rose; but, now it's dead again. Rather than post my broken conf files here, can anyone suggest a nice'n'easy way to get this to work? Many thanks in advance. Ade. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset DECT base provisioning
2007/8/13, Paul Hayes [EMAIL PROTECTED]: It's not currently possible but Siemens are working on new firmware for at least the S450IP model which will support auto-config using http. I'm not sure when it's due for release though. Thanks for the tip ! Directly asking to Siemens ( http://gigaset.siemens.com/shc/0,1935,hq_en_0_11729_rArNrNrNrN,00.html) before posting to this list, was not very helpful (to say the least). How should I track this firmware release ? Should I just check with http://gigaset.siemens.com/shc/0,1935,hq_en_0_123868_rArNrNrNrN_variation%253A-5_pageType%253Adownloads_imagePos%253A0,00.html#content for post V02063 firmware ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through a PAP2
Did you try T.38 ? These PAP2 boxes should be able to benefit from Asterisk T.38 pass through capabilities. You would then have to install a T.38 termination device, such as Linksys 3102 : PSTN Linksys 3102 --- LAN - PAP2 --- Fax machine Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Firefly IAX2 configuration
Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password host=dynamic Then I ran the following: #/usr/sbin/asterisk -cvvv CLIreload But always I get a message at the firefly that an error occured while trying to connect to the network. What else I have to do? By the way: what is the command that I can type it to do tracing on the user [iax2user1] or to do traces on any registeration attempts from the clients? Last thing, if I am outside the console (in unix mode), is there any command from unix I can type it to know if asterisk is running or not? Regards Bilal Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redundancy / Failover
Dears Any one succeeded to make Redundancy / Failover with asterisk 1.4.9 on centos with kernel 2.6.9-55.EL. Can you please send me the documentation link on how to or write down how to . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Application for Home Delivery Restaurants
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kashif Naeem wrote: Hello All We have developed an application for Home Delivery Restaurants using Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If someone is interested then we can provide him more details. - POP up window with caller data containing his/her name, address and transactions history. - In case of new customer, Pop up window with blank form to add customer data and order detail. - Invoice generation and print functionality of Invoice. - Black list a customer if he placed fake order and next time its black list status would show based on his CLI. - Call recording - Sales Analysis URL? Licence? I'm assuming free seeing as this was sent to the Non-Commercial Discussion list. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGyUvYDQNt8rg0Kp4RAodVAJ90MjdlubuVD0Em6ekXXkjWi6uy3gCfVGzu E4u0QbRRxKTG1AvRL5kgUU8= =iiJk -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firefly IAX2 configuration
On Mon, 20 Aug 2007, bilal ghayyad wrote: Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password host=dynamic Then I ran the following: #/usr/sbin/asterisk -cvvv CLIreload But always I get a message at the firefly that an error occured while trying to connect to the network. What else I have to do? Have you checked your firewall? Is it letting UDP data through to the asterisk box on port 4569? By the way: what is the command that I can type it to do tracing on the user [iax2user1] or to do traces on any registeration attempts from the clients? iax2 debug will generate lots of output for you... Last thing, if I am outside the console (in unix mode), is there any command from unix I can type it to know if asterisk is running or not? ps ax | grep asterisk is crude, but visual. Asterisk stores it's PID in /var/run/asterisk.pid, so you could then read that, and check to see if the process with that PID is actually running asterisk. ie. see if /proc/number existis, and if-so, see if it's actually asterisk by reading /proc/number/cmdline or just see if you can connect to it with the rasterisk command ... Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues with Dynanic Users (BUG?)
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to dynamically make
themselves a Queue Member or not.
So, operators call 511, and they should get added to the Queue as a Queue
member.
When users call 510 then, it actually does ring everyone who has called 511.
The problem is when the operator comes to pick up the call. The operator hears
nothing, and the user still hears the Music on Hold. Not only that, but after
about 5 seconds, the operators call gets dropped.
Is there anything that I am doing wrong?
Thanks,
Tim
here are snipits of my config files:
== extensions.conf ==
[default]
exten = 510,1,Answer
exten = 510,2,Queue(techsupport,t)
exten = 511,2,Set(CALLBACKNUM=${CALLERID(number)})
exten = 511,3,AddQueueMember(techsupport)
exten = 511,4,Playback(queue-techsupport-in)
exten = 511,5,Hangup
== queues.conf ==
[techsupport]
music=default
strategy = ringall
timeout = 10
retry = 2
maxlen = 0
announce-frequency = 10
announce-holdtime = yes
== agents.conf ==
[general]
ackcall=no
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Re: [asterisk-users] Queues with Dynanic Users (BUG?)
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote:
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to dynamically make
themselves a Queue Member or not.
So, operators call 511, and they should get added to the Queue as a Queue
member.
When users call 510 then, it actually does ring everyone who has called 511.
The problem is when the operator comes to pick up the call. The operator hears
nothing, and the user still hears the Music on Hold. Not only that, but after
about 5 seconds, the operators call gets dropped.
Is there anything that I am doing wrong?
Thanks,
Tim
here are snipits of my config files:
== extensions.conf ==
[default]
exten = 510,1,Answer
exten = 510,2,Queue(techsupport,t)
exten = 511,2,Set(CALLBACKNUM=${CALLERID(number)})
exten = 511,3,AddQueueMember(techsupport)
exten = 511,4,Playback(queue-techsupport-in)
exten = 511,5,Hangup
== queues.conf ==
[techsupport]
music=default
strategy = ringall
timeout = 10
retry = 2
maxlen = 0
announce-frequency = 10
announce-holdtime = yes
== agents.conf ==
[general]
ackcall=no
Can you also provide output of show queues and show channels? Plus
the logfile of dial to 511.
I'm using QueueAdd after AgentCallbackLogin (trough manager API).
Maybe you need to use AgentCallbackLogin first?
Regards,
Atis
--
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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Re: [asterisk-users] OT Provisioning http-server-enabled devices (Was: Siemens Gigaset DECT base provisioning)
2007/8/13, Eric Chamberlain [EMAIL PROTECTED]: What you describe is doable; we have a number of device configuration wizards. But it is generally easier to use the device's bulk provisioning methods, like https an XML configuration file to the device. The provisioning settings a pretty standard and don't change very often. The problem with using the user web interface is that the manufacturers quite often change the interface with new firmware releases, so you are constantly updating the scripts. I fully agree with this maintenance concern but what do you exactly mean by use the device's bulk provisioning methods, like https an XML configuration file to the device ? For instance, what about products which do not provide provisioning method that deserves to be called bulk provisioning method ? Do you have a trick to escape from using web interface ? Regards -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Olivier *Sent:* Friday, August 10, 2007 3:13 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] OT Provisioning http-server-enabled devices (Was: Siemens Gigaset DECT base provisioning) hello, I would to define and unattended process to configure devices which are http-server-enabled, use DHCP but do not use TFTP-DCHP to configure themselves during boot. Has anyone worked on such subject ? I was thinking of something like : populating configuration file from device web pages (rendering this as generic and flexible as possible) writing a script which reads this file and set each parameter using http writing a script which monitors network environment to trigger previous when certain events occur. All this is not very clear for me, yet. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Thanks Steve and Mitcheloc, in fact i was think in something more obsolet like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, mitcheloc [EMAIL PROTECTED] wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.4. caller ID not working ?
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI to [EMAIL PROTECTED]
As of now, only the _display name_ is being replaced, but not the name.
I tried CALLERID(num) as well CALLERID(number), to the same effect(only
display name being set to number).
Anyone facing similar problems?
Thanks in advance.
- Ben
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Re: [asterisk-users] Nokia cell connected to Asterisk
Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Thanks, Steve Jonathan GF wrote: Thanks Steve and Mitcheloc, in fact i was think in something more obsolet like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, *mitcheloc* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com http://www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.4. caller ID not working ?
Also,
. if I use Remote-party-id header, can it be different from the 'From' URI?
. If yes, how do you achieve this in Asterisk?
. What(From or Remote-party-id) is used by clients to show as the CLI of
the caller?
if I am not mistaken, Remote-party-id is for network elements to confirm
identities of end subscribers.
All corrections and suggestions welcome.
- Ben
Benjamin Jacob wrote:
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI to [EMAIL PROTECTED]
As of now, only the _display name_ is being replaced, but not the name.
I tried CALLERID(num) as well CALLERID(number), to the same effect(only
display name being set to number).
Anyone facing similar problems?
Thanks in advance.
- Ben
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Re: [asterisk-users] Firefly IAX2 configuration
Dear Gordon; Thanks a lot for your email. I need one more tracing tool, how can I know the used port of the IAX on teh Asterisk and wethor the listening on that port is successully done (ready to receive on that port)? About the firewall, actually the client PC and Asterisk on the same LAN (my PC is 192.168.8.2 and Asterisk is 192.168.8.4), the only possible thing is the firewall on the fedora server (Asterisk server), but I am not so friendly with fedora to know how can I check if the firewall on fedora enabled if u can help me (fedora is like redhat). Regards Bilal Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password host=dynamic Then I ran the following: #/usr/sbin/asterisk -cvvv CLIreload But always I get a message at the firefly that an error occured while trying to connect to the network. What else I have to do? Have you checked your firewall? Is it letting UDP data through to the asterisk box on port 4569? By the way: what is the command that I can type it to do tracing on the user [iax2user1] or to do traces on any registeration attempts from the clients? iax2 debug will generate lots of output for you... Last thing, if I am outside the console (in unix mode), is there any command from unix I can type it to know if asterisk is running or not? ps ax | grep asterisk is crude, but visual. Asterisk stores it's PID in /var/run/asterisk.pid, so you could then read that, and check to see if the process with that PID is actually running asterisk. ie. see if /proc/number existis, and if-so, see if it's actually asterisk by reading /proc/number/cmdline or just see if you can connect to it with the rasterisk command ... Gordon Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firefly IAX2 configuration
On Mon, 20 Aug 2007, bilal ghayyad wrote: Dear Gordon; Thanks a lot for your email. I need one more tracing tool, how can I know the used port of the IAX on teh Asterisk and wethor the listening on that port is successully done (ready to receive on that port)? Use netstat -lnveep to list open ports and display the programs using them. About the firewall, actually the client PC and Asterisk on the same LAN (my PC is 192.168.8.2 and Asterisk is 192.168.8.4), the only possible thing is the firewall on the fedora server (Asterisk server), but I am not so friendly with fedora to know how can I check if the firewall on fedora enabled if u can help me (fedora is like redhat). I don't know fedora either, but try: iptables -n -L and it it spews forth lots and lots of lines, then there is local firewalling. You can turn all iptable firewalling off with: iptables --flush iptables --delete-chain but it will restore upon reboot (probably) Whether turning all firewalling off is a good thing or not, is up to you, but as it's on a private LAN, then I'd suggest it's probably OK. Gordon Regards Bilal Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password host=dynamic Then I ran the following: #/usr/sbin/asterisk -cvvv CLIreload But always I get a message at the firefly that an error occured while trying to connect to the network. What else I have to do? Have you checked your firewall? Is it letting UDP data through to the asterisk box on port 4569? By the way: what is the command that I can type it to do tracing on the user [iax2user1] or to do traces on any registeration attempts from the clients? iax2 debug will generate lots of output for you... Last thing, if I am outside the console (in unix mode), is there any command from unix I can type it to know if asterisk is running or not? ps ax | grep asterisk is crude, but visual. Asterisk stores it's PID in /var/run/asterisk.pid, so you could then read that, and check to see if the process with that PID is actually running asterisk. ie. see if /proc/number existis, and if-so, see if it's actually asterisk by reading /proc/number/cmdline or just see if you can connect to it with the rasterisk command ... Gordon Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
On Monday 20 August 2007 8:16:32 pm Atis wrote: On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote: Can you also provide output of show queues and show channels? Plus the logfile of dial to 511. I'm using QueueAdd after AgentCallbackLogin (trough manager API). Maybe you need to use AgentCallbackLogin first? Regards, Atis == Call to 511 == Saved useragent Linksys/SPA3102-3.3.6(GW) for peer 101 == Using TOS bits 0 == Using CoS mark 5 -- Executing [EMAIL PROTECTED]:2] Set(SIP/101-0821ca48, CALLBACKNUM=101) in new stack -- Executing [EMAIL PROTECTED]:3] AddQueueMember(SIP/101-0821ca48, techsupport) in new stack [Aug 20 22:28:26] NOTICE[6238]: app_queue.c:3441 aqm_exec: Added interface 'SIP/101' to queue 'techsupport' -- Executing [EMAIL PROTECTED]:4] Playback(SIP/101-0821ca48, queue-techsupport-in) in new stack -- Executing [EMAIL PROTECTED]:5] Hangup(SIP/101-0821ca48, ) in new stack == Spawn extension (mor, 511, 5) exited non-zero on 'SIP/101-0821ca48' == Outputs == *CLI show channels No such command 'show channels' (type 'help' for help) *CLI show queues No such command 'show queues' (type 'help' for help) *CLI queue show techsupport has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
JR Richardson wrote: Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? I'm interested in putting together a new-user tutorial about DUNDi configuration and setup. There is a lot of great information, setup guides already but the feedback I get is that the current examples are a bit complicated to follow for new users. Your feedback is appreciated. Thanks. JR I just happened to spend some time this weekend messing with DUNDi after hearing the discussion on the asterisk users conference. I would say there is definitely room for improvement in the documentation. I did manage to get it working but there were a few things that would have helped me get moving more quickly: * I took me quite a while (and I'm still not sure I get it all) to understand what exactly a dundi context is. What are best-practices in naming them? Where else does this name get used? Something that showed the relationships between dundi context in the mappings section, the peers section and how it's used (and I mean more than just use a switch= statement) in the dialplan would be helpful. * Stating more clearly that the [mappings] section of dundi.conf determines how OTHER systems map dundi searches in a specific dundi context to extentions.conf contexts and how to connect to them on THIS system. * I had to guess a little bit about how to use dynamic peers. dundi.conf has a register=yes option but it doesn't specify how you told asterisk that it had a dynamic address. Knowing how it's done in IAX and SIP I just copied that syntax and it seemed to work. Also, an example with a dynamic peer would be helpful. I haven't gotten this far in testing, but if a site has a dynamic address how do you set up the IAX channel so the static side can contact the dynamic via IAX? As for your #2 and #3 questions: Once you have a basic understanding of what the components of DUNDi are and how they work I think it's only slightly more complex than setting up an IAX trunk between two systems. Which is all you're really doing anyway with some added features. I think simple examples showing you how to setup a DUNDi cloud with two systems that explained what each part of the config file accomplished would be very helpful to new users. A HOWTO that gives me a cookie cutter config file and says put host A address here, put host B address here... may get me a working setup but I still don't have any idea how to expand it without understanding it. My 2 cents anyway. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote: On Monday 20 August 2007 8:16:32 pm Atis wrote: On 8/20/07, Tim Groeneveld [EMAIL PROTECTED] wrote: Can you also provide output of show queues and show channels? Plus the logfile of dial to 511. I'm using QueueAdd after AgentCallbackLogin (trough manager API). Maybe you need to use AgentCallbackLogin first? Regards, Atis == Call to 511 == Saved useragent Linksys/SPA3102-3.3.6(GW) for peer 101 == Using TOS bits 0 == Using CoS mark 5 -- Executing [EMAIL PROTECTED]:2] Set(SIP/101-0821ca48, CALLBACKNUM=101) in new stack -- Executing [EMAIL PROTECTED]:3] AddQueueMember(SIP/101-0821ca48, techsupport) in new stack [Aug 20 22:28:26] NOTICE[6238]: app_queue.c:3441 aqm_exec: Added interface 'SIP/101' to queue 'techsupport' -- Executing [EMAIL PROTECTED]:4] Playback(SIP/101-0821ca48, queue-techsupport-in) in new stack -- Executing [EMAIL PROTECTED]:5] Hangup(SIP/101-0821ca48, ) in new stack == Spawn extension (mor, 511, 5) exited non-zero on 'SIP/101-0821ca48' == Outputs == *CLI show channels No such command 'show channels' (type 'help' for help) *CLI show queues No such command 'show queues' (type 'help' for help) *CLI queue show techsupport has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers Ok, i just noticed that you are running trunk. Probably you should write to asterisk-dev then. Seems that agent get's added correctly. So you could try to view (and post us) log of call to queue, maybe it says something. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
I wouldn't exactly say that it is too difficult but that the target audience for the default examples is not the average person/entity that could make use of the power inherent with DUNDi. When an average * user/admin wants to use DUNDi they will want to start out small and local rather than worry about all of the intricacies of the e164 standard. It is much easier, in my opinion, to learn the power of DUNDi on a simple level and scale that up to a more globally connected platform. I'd say that duni.conf is a reference, and you expect it to be an introductory document. A reference should be comprehensive. It is best used after you've grasped the basic concepts, and together with a text search. Asterisk's sample configuration files actually serve a role of a reference. The config files can be both a reference and an introduction. Look at sip.conf. Most of the examples in that file are relatively simple, what you would expect for a beginner to set up most of the time. There are also some more complex examples in that file. Lastly, the sip.conf file has a good section that explains pretty much any option that could be used in sip.conf. We should strive to make all of the conf files similar to sip.conf and iax.conf. I don't disagree with you that a separate intro document is needed but there is no reason that the conf files could not serve a broader purpose. Matthew Brothers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rewriting the From and Subject from voicemail for a MMS Message to a Cell Phone - like visual voicemail
On 8/19/07, Steve Turner [EMAIL PROTECTED] wrote:
I would like to send Multimedia Messaging (MMS) email (gateway) to my
cell
phone and have the from and subject be the callerid/calleridnam
information
from the voice mail message.
voicemail.conf lets you change the from and subject line, and has
replacement tokens for ${VM_CALLERID}, ${VM_CIDNUM} and ${VM_CIDNAME}.
What are you trying to achieve that use of emailfrom, emailsubject and
attach=yes in voicemail conf won't do?
--
j.
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[asterisk-users] Got SUBSCRIBE for extension...., but there is no hint for that extension.
Hi all, I am seeing the following messages on my asterisk cli: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.0.158, but there is no hint for that extension. I dont know what it means. I believe it has something to do with realtime extensions or hints. i know about realtime extensions which i am not using. So what r hints? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
On Mon, Aug 20, 2007 at 09:26:00AM -0400, Matthew Brothers wrote: I wouldn't exactly say that it is too difficult but that the target audience for the default examples is not the average person/entity that could make use of the power inherent with DUNDi. When an average * user/admin wants to use DUNDi they will want to start out small and local rather than worry about all of the intricacies of the e164 standard. It is much easier, in my opinion, to learn the power of DUNDi on a simple level and scale that up to a more globally connected platform. I'd say that duni.conf is a reference, and you expect it to be an introductory document. A reference should be comprehensive. It is best used after you've grasped the basic concepts, and together with a text search. Asterisk's sample configuration files actually serve a role of a reference. The config files can be both a reference and an introduction. Look at sip.conf. Most of the examples in that file are relatively simple, what you would expect for a beginner to set up most of the time. There are also some more complex examples in that file. Lastly, the sip.conf file has a good section that explains pretty much any option that could be used in sip.conf. We should strive to make all of the conf files similar to sip.conf and iax.conf. It explains the configuration file. But it does not explain the SIP channel. And it is very very long. way too long to be useful for a beginner. Also, if you have NAT issues, what makes you think you should actually have a look in the section for media handling. What exactly is the meaning of path there? What does user mean? What does peer mean? A simple text-search in the document is not useful enough, as those two words appear in different contexts as well. This file has a lots of useful information. But it will not be useful enough to a novice admin without a nicer introduction. (But then again, if anybody wishes to write something, I won't say no) I don't disagree with you that a separate intro document is needed but there is no reason that the conf files could not serve a broader purpose. One obvious reason: it gets in the way of the original role as sample config files. If you have a huge sip.conf , you can't manage it. (even if you heard if 'grep -v ^; filename.conf') -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disabling Asterisk Authentication
Hello, I have a small LAN network connected through an Asterisk Server. When I try to make a call between two of the user pc's on this network I get a 401 Unauthorized error. Would anyone know how to remove the Asterisk Authorization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ; do not edit this file, this is an auto-generated file by freepbx ; all modifications must be done from the web gui [201] type=friend secret=201 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device 201 [202] type=friend secret=202 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device 202 Thanks very much, Denis Kutman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 asterisk servers, how to connect them together?
the following link show more than one methods to connect 2 asterisk servers: http://www.voip-info.org/wiki-Asterisk+-+dual+servers On 8/19/07, Ade Vickers [EMAIL PROTECTED] wrote: Panic over... I have a weird network problem (now solved), whereby incoming packets arrived directly to the Asterisk box (eth1); but outgoing packets attempted to leave via the LAN (eth0)... solved it by sending the IAX packets thru the firewall at both ends of the connection (i.e. binding IAX to the LAN address instead of the WAN address). Freaky? You betcha... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ade Vickers Sent: 18 August 2007 23:47 To: asterisk-users@lists.digium.com Subject: [asterisk-users] 2 asterisk servers, how to connect them together? Hi... I have what is, I am sure, a relatively common straightforward problem (no, NOT that kind of problem!)... I'm trying to hook two asterisk servers together so I can make a distributed PBX. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.0/960 - Release Date: 18/08/2007 15:48 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Limits
well i have mentioned earlier that it happens to only one user. all of the other users limits are working fine. In asterisk 1.4.0, this was a bug as it happened to every user. But then i upgraded to 1.4.2 and it was gone. It was working fine since then but recently this problem again showed up for a single user. I dont understand that, if this bug is still around then why isnt it bothering other users? On 8/17/07, Remi Quezada [EMAIL PROTECTED] wrote: I think its an Asterisk bug, call-limits stopped working for me once I upgraded from 1.2.16 to 1.2.18. There is a bug opened for it, but the issue hasn't been resolved yet. Here is the link: http://bugs.digium.com/view.php?id=9794 -Remi Ira wrote: At 06:37 AM 8/17/2007, you wrote: Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
Tim Groeneveld wrote:
I am running r79979 of Asterisk Trunk, and I am having problems trying to use
app_queue.so.
I want to use the extension 510 to be a line where users can call technical
support.
Extensions 511 and 512 are used by the operators to dynamically make
themselves a Queue Member or not.
So, operators call 511, and they should get added to the Queue as a Queue
member.
When users call 510 then, it actually does ring everyone who has called 511.
The problem is when the operator comes to pick up the call. The operator
hears
nothing, and the user still hears the Music on Hold. Not only that, but after
about 5 seconds, the operators call gets dropped.
Is there anything that I am doing wrong?
Thanks,
Tim
here are snipits of my config files:
== extensions.conf ==
[default]
exten = 510,1,Answer
exten = 510,2,Queue(techsupport,t)
exten = 511,2,Set(CALLBACKNUM=${CALLERID(number)})
exten = 511,3,AddQueueMember(techsupport)
exten = 511,4,Playback(queue-techsupport-in)
exten = 511,5,Hangup
== queues.conf ==
[techsupport]
music=default
strategy = ringall
timeout = 10
retry = 2
maxlen = 0
announce-frequency = 10
announce-holdtime = yes
== agents.conf ==
[general]
ackcall=no
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Remove the Answer() before the call to Queue(). See if that corrects the
problem.
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Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
On 8/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: (But then again, if anybody wishes to write something, I won't say no) So why all the verbiage? JR offered a valuable service to the community, I see no downside to this. If anyone doesn't care for the idea they can just ignore it. A lot of people including me will applaud his efforts. Over the years I've read these mailing lists, many people have done a lot for the community and the state of the art of asterisk without writing a single line of code. I think that's great and I'm sure Mark Spencer would agree. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - IMAP voicemail statistics
Hi, Has anyone experienced a tool providing system administrators with IMAP voicemail statistics ? The main usage is know the amount of time between the moment a message is dropped in voicemail and the moment this message is read (heard). I guess this is required to help management to pin point users or services who never reply. Do you know any IMAP server offering such capability ? Which keyword shall I enter to check if this feature is supported by a given email server ? Best regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as ISDN PRI Proxy
I have a costumer with a Siemens PBX installed, and I would like to setup a Asterisk system that would act as a kind of Proxy between the Siemens PBX and the operator network. The current setup is: Siemens PBX 2*PRI - Operator what I want is: Siemens PBX 2*PRI - Asterisk BOX - Operator For the Siemens PBX the Asterisk Box would be a standard Telephony Operator, and the Asterisk box would either route the calls normally, or would route them via another system via SIP or IAX. I need to know if this is possible, and what kind of hardware do I need on the Asterisk Box to do this. I know I'll need some PRI cards to connect to the Operator, but do those cards allow me to masquerade as a Operator to the Siemens PBX? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . - signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Limits
Well only certain situations expose this bug. I am able to reproduce this bug in two instances. One is with the Adtran Total Access 900 series when it receives a fax call it sends a INVITE to the Asterisk. When Asterisk receives this INVITE it changes the call from peer to user, so by the end of the call, the call counter for user gets updated not the peer (details in the bug). The other time is when the peer/user does a re-invite with another asterisk server, same thing happens. So I ended up disabling call limits overall since it wasn't too much of a big deal for me, perhaps other people have disabled it also. -Remi Rizwan Hisham wrote: well i have mentioned earlier that it happens to only one user. all of the other users limits are working fine. In asterisk 1.4.0, this was a bug as it happened to every user. But then i upgraded to 1.4.2 and it was gone. It was working fine since then but recently this problem again showed up for a single user. I dont understand that, if this bug is still around then why isnt it bothering other users? On 8/17/07, *Remi Quezada* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I think its an Asterisk bug, call-limits stopped working for me once I upgraded from 1.2.16 to 1.2.18. There is a bug opened for it, but the issue hasn't been resolved yet. Here is the link: http://bugs.digium.com/view.php?id=9794 -Remi Ira wrote: At 06:37 AM 8/17/2007, you wrote: Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? It's not just you, it happens to my wife too. No rhyme or reason I can see, I just try to restart asterisk occasionally so it doesn't get that far. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com http://www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as ISDN PRI Proxy
On 8/20/07, Gustavo Felisberto [EMAIL PROTECTED] wrote: I have a costumer with a Siemens PBX installed, and I would like to setup a Asterisk system that would act as a kind of Proxy between the Siemens PBX and the operator network. The current setup is: Siemens PBX 2*PRI - Operator what I want is: Siemens PBX 2*PRI - Asterisk BOX - Operator This is not unusual. For the Siemens PBX the Asterisk Box would be a standard Telephony Operator, and the Asterisk box would either route the calls normally, or would route them via another system via SIP or IAX. I need to know if this is possible, and what kind of hardware do I need on the Asterisk Box to do this. I know I'll need some PRI cards to connect to the Operator, but do those cards allow me to masquerade as a Operator to the Siemens PBX? look at pri_cpe vs pri_net -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
On 8/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: In an attempt to understand why there are no better docs inside asterisk. Well, we're all on the same page then :) My opinion, summed up into a sentence would be that the people who create the code have *mostly* commented the main conf files very well. So well (oh heck, I said one sentence) that they are filled with stuff you probably won't use as in sip.conf or in the case of the last features.conf I looked at, totally incomprehensible. Documentation is obviously the process of documenting and will often be terse and to the point. OTH, I think the words tutorial, guide, manual, how-to and cookbook should be most welcome! I hope a thousand JR's write a thousand tutorials on a thousand aspects of asterisk! We'll get them all talking on http://www.AsteriskUsersConference.org /r ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cdr reports
I am trying to figure out how long a caller waited in queue for someone to answer versus how long they stayed on the phone after the answer. I am assuming that the duration is the total talk time and that the billsecs are the total time in queue. is this correct? or should i be deducting the billsecs from the duration to get this number? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SpanDSP/TxFAX FAX Status
Hi List, I wonder that how I can check that FAX is delivered successfully or not, in my dialplan while using TxFAX. Which variable contain fax status? as PHASEESTATUS and PHASEESTRING in Callweaver. Regards Nasir Iqbal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cdr reports
On Mon, 2007-08-20 at 11:45 -0500, Jordan Novak wrote: I am trying to figure out how long a caller waited in queue for someone to answer versus how long they stayed on the phone after the answer. I am assuming that the duration is the total talk time and that the billsecs are the total time in queue. is this correct? or should i be deducting the billsecs from the duration to get this number? That information you need to extract from queue_log and not from the CDR. You need something like Queuemetrics to give you comprehensive reports on queue and agent activity. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de TecnologÃa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cdr reports
On Mon, 2007-08-20 at 11:45 -0500, Jordan Novak wrote: I am trying to figure out how long a caller waited in queue for someone to answer versus how long they stayed on the phone after the answer. I am assuming that the duration is the total talk time and that the billsecs are the total time in queue. is this correct? or should i be deducting the billsecs from the duration to get this number? I HOPE you have the two reversed. Usually, the duration is the total time from the moment Asterisk picked up the incoming call, to the time the conversation ended. The billsecs is usually smaller-- the time from the moment the callee answered the ringing phone to the end of the conversation. So, the billsec field is usually SMALLER than the duration. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unsuscribe
From: Hans Feringa [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 99 bottles of beer
Date: Fri, 17 Aug 2007 15:10:16 +0200 (CEST)
I dialed it, but I am still thirsty. ;-)
On Thu, 2007-08-16 at 19:38 -0600, Steve Murphy wrote:
On Thu, 2007-08-16 at 07:56 -0400, Russell Bryant wrote:
Gordon Henderson wrote:
; *99:
; 99 bottles of beer on the wall.
exten = *99,1,Noop(99 Bottles of beer on the wall)
exten = *99,n,Answer()
exten = *99,n,Set(bottles=99)
exten = *99,n(loop),Noop(There are ${bottles} bottles of beer on
the wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,Noop(Take one done and pass it round and there's)
exten = *99,n,Set(bottles=$[${bottles}-1])
exten = *99,n,Noop(${bottles} bottles of beer on the wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,GotoIf($[${bottles} 0]?loop)
exten = *99,n,Noop(We're out of beer!)
exten = *99,n,Hangup()
Too much dial plan mashing this morning and I rememberd this site:
http://99-bottles-of-beer.net/
And now, in AEL! (This is untested, I just wanted to see how it
would
look.)
context silly {
*99 = {
NoOp(99 Bottles of beer on the wall);
Answer();
bottles=99;
while (${bottles} 0) {
NoOp(${bottles} bottles of beer on the wall, ${bottles} bottles
of beer);
SayNumber(${bottles});
NoOp(Take one down, pass it around);
bottles=${bottles} - 1;
NoOp(${bottles} bottles of beer on the wall);
}
NoOp(We're out of beer!);
Hangup();
}
}
Lol, Well done, Russell!
How about this one: from an extensions.conf that someone posted on the
internet, I think, and I converted to AEL; I'm sorry, but I can't find
the original author.
(If anybody can find his post, I'd love to give him credit.) I did test
this out,
and it works; just put a call to the macro ( guessgame(); ) in an
extension in your dialplan
macro guessgame()
{
startpoint:
while (1)
{
Playback(guessit/intro);
set(GUESS=);
GUESS=${EPOCH}%9;
Set(TIMEOUT(digit)=3);
Set(TIMEOUT(response)=5);
while (1)
{
Read(NUMBER,guessit/input_number,1);
Verbose(Got ${NUMBER} from Read);
if( ${NUMBER} = * || ${NUMBER} = # ||
${NUMBER} = )
{
Playback(guessit/thatsnotanumber);
}
else if (${NUMBER} = ${GUESS})
{
Playback(guessit/win);
break; // the only way out of this loop!
}
else if (${NUMBER} ${GUESS})
{
Playback(guessit/less);
}
else if (${NUMBER} ${GUESS})
{
Playback(guessit/more);
}
else /* what other stuff can the user enter than a
number, #,
nothing, or * ? */
{
Playback(guessit/thatsnotanumber);
}
}
/* You get here after a successful guess */
Wait(.5);
Read(AGAIN,guessit/playagain,1);
if (${AGAIN} != 1)
break;
}
Playback(guessit/goodbye);
return;
catch t
{
playback(guessit/goodbye);
return;
}
catch i
{
playblack(invalid);
}
}
murf
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Hey murf,
here is the link for the credit,
http://www.voipphreak.ca/archives/358-Asterisk-Howto-Numbers-Guessing-Game.html
its also in the wiki examples.
http://www.voip-info.org/wiki/view/AEL+Example+Snippets
db
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[asterisk-users] Setting caller ID on outgoing calls.
Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the outgoing number instead of the incoming number? I believe the answer is (was) yes. New twist, does it matter what the destination media is? Meaning, the call would be coming in on a T1, going out on a T1, but ending on a POTS line (which supports caller ID). Thanks for understanding. joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cdr reports
Steve Murphy wrote: So, the billsec field is usually SMALLER than the duration. Usually? Are there situations when the duration is smaller than billsec? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Asterisk Authentication
comment the line secret=201 or 202 Then it'll not ask for 401Autherization. Regards, Kesh [EMAIL PROTECTED] wrote: Hello, I have a small LAN network connected through an Asterisk Server. When I try to make a call between two of the user pc's on this network I get a 401 Unauthorized error. Would anyone know how to remove the Asterisk Authorization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ; do not edit this file, this is an auto-generated file by freepbx ; all modifications must be done from the web gui [201] type=friend secret=201 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device 201 [202] type=friend secret=202 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device 202 Thanks very much, Denis Kutman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Kesh Lets change the future...lets change the world. - Pinpoint customers who are looking for what you sell. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Has anyone ever tried using a Nokia phone with SIP client as channel for Asterisk? I mean i would like to receive calls to the mobile on asterisk and use the Nokia phone to place calls to cell destinations. I have enough Nokia E60's to do that and it would circumvent the need for chan_bluetooth or something similar!! :) On Mon, 20 Aug 2007, Steve Totaro wrote: Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Thanks, Steve Jonathan GF wrote: Thanks Steve and Mitcheloc, in fact i was think in something more obsolet like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, *mitcheloc* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com http://www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cdr reports
On 8/20/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Mon, 2007-08-20 at 11:45 -0500, Jordan Novak wrote: I am trying to figure out how long a caller waited in queue for someone to answer versus how long they stayed on the phone after the answer. I am assuming that the duration is the total talk time and that the billsecs are the total time in queue. is this correct? or should i be deducting the billsecs from the duration to get this number? That information you need to extract from queue_log and not from the CDR. You need something like Queuemetrics to give you comprehensive reports on queue and agent activity. The problem with queue_log is - that it is not stored in mysql :p For me this works great: On entering queue, call ResetCDR() - if there are some prompts/IVRs before.. I think, this also resets answer status, so you will have new answer status for agent picking up in queue's CDR. Queue will have one CDR, and each ringed agent will have another (and some more in 1.4). Then in queue's CDR you will have billsec - conversation time, and duration - total time while connected to queue. So - waiting time is billsec-duration. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting caller ID on outgoing calls.
I did post recently, under another subject line. But would appreciate some response, as some are telling a client that this is not possible. joe a. On 8/20/2007 at 1:57 PM, Joe acquisto [EMAIL PROTECTED] wrote: Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the outgoing number instead of the incoming number? I believe the answer is (was) yes. New twist, does it matter what the destination media is? Meaning, the call would be coming in on a T1, going out on a T1, but ending on a POTS line (which supports caller ID). Thanks for understanding. joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting caller ID on outgoing calls.
Am Montag, den 20.08.2007, 13:57 -0400 schrieb Joe acquisto: Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the outgoing number instead of the incoming number? I believe the answer is (was) yes. I seem to remember there is an option to the Dial command, possibly o. Check on the voip-info.org wiki. Maybe this is not true for 1.4 anymore, no idea. There seems to be a difference between transfer with or without speaking to the callee first, and most probably transfers made from the phone features (i.e. transfer button on some phones) will not allow to send the original caller ID. New twist, does it matter what the destination media is? Meaning, the call would be coming in on a T1, going out on a T1, but ending on a POTS line (which supports caller ID). Not the media(SIP/T1) is the problem for outgoing caller ID, but the provider/carrier. For SIP, Zap... devices connected to your asterisk as their server, you can of course send any callerid you want. As soon as you have to hand over the call to any provider, be it SIP, T1, ISDN, IAX,... you have to check wether they allow to set any ID, to only set your own IDs (which are assigned to the outgoing line as incoming numbers) or if they allow doing any CALLERID changing at all. Some providers do not even allow to block the own number, outgoing (for example, afaik, 11 SIP in Germany), others allow to set only numbers assigned to the phone line in question or block calls (without special agreements this seems to be the standard setup for PRI, ISDN and analogue lines in Germany) - others allow to send any number if it is valid (where valid means the provider's idea of a phone number, it seems) - sipgate.de seems to allow to set nearly anything starting with +49. From what I learned about North American whereabouts the situation seems to be similar - business providers seem to be willing to offer more options to their customers, check with your providers. BR HTH Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
On the SIP side of things, we have a how-to guide for the Nokia E series and Asterisk. http://voxilla.com/voxilla-stories/voxilla-how-to-guides/using-the-nokia-e-series-phones-with-asterisk-865.html -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan GF Sent: Monday, August 20, 2007 4:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nokia cell connected to Asterisk Thanks Steve and Mitcheloc, in fact i was think in something more obsolet like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, mitcheloc [EMAIL PROTECTED] wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com http://www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Using the phone itself as a GSM-SIP gateway is not possible with the native VoIP application, but it looks like it should be possible with a custom application for the phone. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Remco Barendse Sent: Monday, August 20, 2007 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nokia cell connected to Asterisk Has anyone ever tried using a Nokia phone with SIP client as channel for Asterisk? I mean i would like to receive calls to the mobile on asterisk and use the Nokia phone to place calls to cell destinations. I have enough Nokia E60's to do that and it would circumvent the need for chan_bluetooth or something similar!! :) On Mon, 20 Aug 2007, Steve Totaro wrote: Well chan_bluetooth is really amazing (especially if your phone does not support SIP). You connect your phone via bluetooth to your asterisk box and it becomes a channel type. You can use it as an extension(FXS) or a phone line (FXO). I believe you can send and receive SMS through the phone/Asterisk as well. Chan_bluetooth README is in the asterisk-addons trunk and gives you basic instruction on setting it up. You get several added pieces of functionality with this setup. SMS send and receive through your phone using Asterisk?, FXO failover or LCR, FXS where your cell phone becomes an extension. Thanks, Steve Jonathan GF wrote: Thanks Steve and Mitcheloc, in fact i was think in something more obsolet like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, *mitcheloc* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com- - asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com http://www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing
[asterisk-users] Realtime Queue Members
Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general] realtime_family=queue_members persistentmembers = yes autofill = yes monitor-type = MixMonitor [queue2280] music = default strategy = roundrobin timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = no joinempty = yes extconfig.conf: [settings] queue_members=mysql,ASTERISK,queue_member_table MYSQL: [EMAIL PROTECTED]:/etc/asterisk# mysql -u myuser -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 7 to server version: 5.0.24a-Debian_9ubuntu2-log Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use ASTERISK; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select * from queue_member_table; ++---+-+ | queue_name | interface | penalty | ++---+-+ | queue2280 | SIP/2224 | 1 | | queue2280 | SIP/2223 | 1 | | queue2280 | SIP/ | 2 | ++---+-+ 3 rows in set (0.00 sec) I don't see any log info for mysql, except when I manually enter the info above. I've stopped an restarted * many times. I've even tried this on two separate boxes and I get the same thing. sipeers and voicemail work, but queue members does not. Any idea? I am running 1.4.10.1. Thanks. Peder ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Subscribe/Notify MWI not working for non-numeric accounts w/X-Lite
John C. Wolosuk Jr. [EMAIL PROTECTED] writes: ok this is a wired problem. when i use X-Lite - after i register with asterisk X-lite sends a subscribe/notify request to asterisk to determine if the account has any messages waiting. if i create a sip.conf account using: user 12345 with a voicemail box 12345 - MWI works user jwolosuk with a voicemail box 12345 MWI fails (gets a 404 not found upon a subscribe) does anyone have a clue why this doesn't work or what i need to set in order for this to work? feedback is appreciated, Huh, not sure. Haven't used X-Lite in a long time, but maybe put a regexten=12345 for user jwolosuk? -- Kyle Sexton ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Subscribe/Notify MWI not working for non-numeric accounts w/X-Lite
John C. Wolosuk Jr. wrote: ok this is a wired problem. when i use X-Lite - after i register with asterisk X-lite sends a subscribe/notify request to asterisk to determine if the account has any messages waiting. if i create a sip.conf account using: user 12345 with a voicemail box 12345 - MWI works user jwolosuk with a voicemail box 12345 MWI fails (gets a 404 not found upon a subscribe) does anyone have a clue why this doesn't work or what i need to set in order for this to work? feedback is appreciated, Do you have [EMAIL PROTECTED] under [jwolosuk] in your sip.conf? -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
Kyle Sexton wrote: Maybe we can convince Digium to have an indemnification program for people who purchase the business edition! :) This is already in place. Asterisk Business Edition is delivered under a traditional commercial software license that includes warranty protection and patent indemnification, among other benefits. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager Proxy - Still required?
On 8/17/07, Andrew Ruthven [EMAIL PROTECTED] wrote: Hi, With more recent version of v1.2 and with v1.4 are things like the AstManProxy still recommended if you want to have a bunch of applications talking directly to Asterisk? If you're looking to have a number of clients monitor events, etc, I'd say that having a proxy in between is still a good thing. The performance of the manager itself is greatly improved since before 1.2 but there are still ongoing (albeit random, sporadic) issues with cpu race and huge memory allocations that still need to get resolved. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.2.20 echo cancelling problem
On my Asterisk installation, I've had to roll back to Zaptel 1.2.19. When I use 1.2.20, I get very bad echo problems. It seems to work OK if I use a quieter-than-normal speaking voice, but at a sufficient sound level, the echo breaks through and then never goes away. The problem goes away if I revert to 1.2.19. Hardware in question: Digium TE110P, Adtran TA750 channel bank, Adtran 1175407L2 FXO card. The phone was on a TDM400P FXS port. Is there something else I need to tweak in 1.2.20? Russ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue Members
Peder @ NetworkOblivion wrote: Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general] realtime_family=queue_members persistentmembers = yes autofill = yes monitor-type = MixMonitor [queue2280] music = default strategy = roundrobin timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = no joinempty = yes extconfig.conf: [settings] queue_members=mysql,ASTERISK,queue_member_table MYSQL: [EMAIL PROTECTED]:/etc/asterisk# mysql -u myuser -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 7 to server version: 5.0.24a-Debian_9ubuntu2-log Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use ASTERISK; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select * from queue_member_table; ++---+-+ | queue_name | interface | penalty | ++---+-+ | queue2280 | SIP/2224 | 1 | | queue2280 | SIP/2223 | 1 | | queue2280 | SIP/ | 2 | ++---+-+ 3 rows in set (0.00 sec) I don't see any log info for mysql, except when I manually enter the info above. I've stopped an restarted * many times. I've even tried this on two separate boxes and I get the same thing. sipeers and voicemail work, but queue members does not. Any idea? I am running 1.4.10.1. Thanks. Peder ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There is no queue_members file, asterisk doesnt know hat you are talking about, you would have to #include queue_members from inside that queue definition. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues with Dynanic Users (BUG?)
On Tuesday 21 August 2007 12:32:12 am Mark Michelson wrote: When users call 510 then, it actually does ring everyone who has called 511. The problem is when the operator comes to pick up the call. The operator hears nothing, and the user still hears the Music on Hold. Not only that, but after about 5 seconds, the operators call gets dropped. Is there anything that I am doing wrong? Remove the Answer() before the call to Queue(). See if that corrects the problem. No, that did not help at all. Maybe I should use AgentLoginCallback? Thanks a million, Tim Groeneveld signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Subscribe/Notify MWI not working for non-numeric accounts w/X-Lite
yep. [EMAIL PROTECTED] to be exact. it's the same in both configs, the essentially the only things i changed is the [name] username= from 12345 to jwolosuk. i should note my version is 1.4.9 and i am serving the configs via asterisk real time. --- John C. Wolosuk Jr. Unix/Linux Systems Administrator Academic Computing Communications Center University of Illinois @ Chicago E-Mail: jwolosuk at uic dot edu --- Trevor Peirce wrote: John C. Wolosuk Jr. wrote: ok this is a wired problem. when i use X-Lite - after i register with asterisk X-lite sends a subscribe/notify request to asterisk to determine if the account has any messages waiting. if i create a sip.conf account using: user 12345 with a voicemail box 12345 - MWI works user jwolosuk with a voicemail box 12345 MWI fails (gets a 404 not found upon a subscribe) does anyone have a clue why this doesn't work or what i need to set in order for this to work? feedback is appreciated, Do you have [EMAIL PROTECTED] under [jwolosuk] in your sip.conf? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Subscribe/Notify MWI not working for non-numeric accounts w/X-Lite
no dice. :-( --- John C. Wolosuk Jr. Unix/Linux Systems Administrator Academic Computing Communications Center University of Illinois @ Chicago E-Mail: jwolosuk at uic dot edu --- Kyle Sexton wrote: John C. Wolosuk Jr. [EMAIL PROTECTED] writes: ok this is a wired problem. when i use X-Lite - after i register with asterisk X-lite sends a subscribe/notify request to asterisk to determine if the account has any messages waiting. if i create a sip.conf account using: user 12345 with a voicemail box 12345 - MWI works user jwolosuk with a voicemail box 12345 MWI fails (gets a 404 not found upon a subscribe) does anyone have a clue why this doesn't work or what i need to set in order for this to work? feedback is appreciated, Huh, not sure. Haven't used X-Lite in a long time, but maybe put a regexten=12345 for user jwolosuk? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue Members
Anthony Francis wrote: There is no queue_members file, asterisk doesnt know hat you are talking about, you would have to #include queue_members from inside that queue definition. Huh? How is including a file going to make realtime access the queue_members database via mysql? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.2.20 echo cancelling problem
Russ Price wrote: On my Asterisk installation, I've had to roll back to Zaptel 1.2.19. When I use 1.2.20, I get very bad echo problems. You should be trying 1.2.21.1 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 http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue Members
I think that revision 80086 in the 1.4 subversion branch would fix this. Julian. Peder @ NetworkOblivion wrote: Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general] realtime_family=queue_members persistentmembers = yes autofill = yes monitor-type = MixMonitor [queue2280] music = default strategy = roundrobin timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = no joinempty = yes extconfig.conf: [settings] queue_members=mysql,ASTERISK,queue_member_table MYSQL: [EMAIL PROTECTED]:/etc/asterisk# mysql -u myuser -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 7 to server version: 5.0.24a-Debian_9ubuntu2-log Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use ASTERISK; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select * from queue_member_table; ++---+-+ | queue_name | interface | penalty | ++---+-+ | queue2280 | SIP/2224 | 1 | | queue2280 | SIP/2223 | 1 | | queue2280 | SIP/ | 2 | ++---+-+ 3 rows in set (0.00 sec) I don't see any log info for mysql, except when I manually enter the info above. I've stopped an restarted * many times. I've even tried this on two separate boxes and I get the same thing. sipeers and voicemail work, but queue members does not. Any idea? I am running 1.4.10.1. Thanks. Peder ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE405/TE410P help updating from 1.0 to 1.4
I have a TE405/TE410P card that was working on 1.0.X I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to 1.4.5 and libpri. I copied all the zaptel and zapata and extensions.conf files from 1.0 I did update extensions.conf from 1.0 to 1.4 commands. I cannot get the card to work in 1.4.10. AHHH! I see with zttool that the T1 is in Green, I see calls coming in as the bits go high on channel 8, zaptel doesnt respond so it tries channel 7 then gives up. Any ideas what this might be??? zaptel modules load, asterisk loads. ztcfg gives correct reply everything looks good just not working. jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing Variables to Voicemail's Email Notification
Is there a way, other then recoding the entire voicemail application, to
pass dialplan variables to the voicemail application and to the email
notifications of new voicemail.
For example in our small tech support queue i would like to pass the ticket
number with the email notification that a new support voicemail was left.
I've tried simply replacing the ${VM_WHATEVER} w/ the actual variable name
inside the voicemail.conf file, I've also tried setting the VM variables
directly before the voicemail application call in the dial plan... both of
these fail.
Anyone else know of another way?
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Re: [asterisk-users] Passing Variables to Voicemail's Email Notification
While I don't have an answer on how to access channel variables from
voicemail.conf, for the problem you mention this should help.
Change CALLERID(name) to your ticket number and then use VM_CIDNAME in
the subject line.
If you don't want to lose the original CIDNAME then just add your
ticket number like this:
Set(CALLERID(name)=${CALLERID(name)} TICKETNUMBER:12345)
On 8/20/07, 0xception [EMAIL PROTECTED] wrote:
Is there a way, other then recoding the entire voicemail application, to
pass dialplan variables to the voicemail application and to the email
notifications of new voicemail.
For example in our small tech support queue i would like to pass the ticket
number with the email notification that a new support voicemail was left.
I've tried simply replacing the ${VM_WHATEVER} w/ the actual variable name
inside the voicemail.conf file, I've also tried setting the VM variables
directly before the voicemail application call in the dial plan... both of
these fail.
Anyone else know of another way?
___
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Re: [asterisk-users] Passing Variables to Voicemail's Email Notification
After rethinking.
I'm not sure if this works, but please report back after testing.
The idea would be that the CIDNAME should not be in the subject just
the ticket number, and the ticket number should not be in the email
body just the CIDNAME.
Please try the following and report back.
exten = _X.,1,Set(BLANKS= );actual 15 spaces, since
CIDName on PSTN should never be longer, and should realy be padded
with blank spaces.
exten =
_X.,n,Set(CALLERID(name)=${CALLERID(name)}${BLANKS:${LEN(${CALLERID(name)})}})
;the above just pads the CIDNAME with blanks so you know for sure it's
at least 15 char long, yes I know if the len of cidname is longer than
blanks then blah.
exten = _X.,n,Set(CALLERID(name)=${CALLERID(name):0:15}=TicketNum:1234)
;this makes sure that it is not longer than 15 plus the ticketnumber.
exten = _X.,n,Voicemail(blah)
In voicemail.conf
emailsubject=${VM_CIDNAME:15}
If this should work then the subject should be: TicketNum:1234
emailbody=New voicemail from ${VM_CIDNAME:0:15} balh.
Again, I'm not sure this will work, please test and report back.
Thank you
On 8/20/07, C F [EMAIL PROTECTED] wrote:
While I don't have an answer on how to access channel variables from
voicemail.conf, for the problem you mention this should help.
Change CALLERID(name) to your ticket number and then use VM_CIDNAME in
the subject line.
If you don't want to lose the original CIDNAME then just add your
ticket number like this:
Set(CALLERID(name)=${CALLERID(name)} TICKETNUMBER:12345)
On 8/20/07, 0xception [EMAIL PROTECTED] wrote:
Is there a way, other then recoding the entire voicemail application, to
pass dialplan variables to the voicemail application and to the email
notifications of new voicemail.
For example in our small tech support queue i would like to pass the ticket
number with the email notification that a new support voicemail was left.
I've tried simply replacing the ${VM_WHATEVER} w/ the actual variable name
inside the voicemail.conf file, I've also tried setting the VM variables
directly before the voicemail application call in the dial plan... both of
these fail.
Anyone else know of another way?
___
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[asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing Variables to Voicemail's Email Notification
AHH lol i can't believe i didn't see/think of that :) thanks .. it's a quick
hack but it works for what i need right now. Maybe this can be a feature
request for the voicemail app
On 8/20/07, C F [EMAIL PROTECTED] wrote:
While I don't have an answer on how to access channel variables from
voicemail.conf, for the problem you mention this should help.
Change CALLERID(name) to your ticket number and then use VM_CIDNAME in
the subject line.
If you don't want to lose the original CIDNAME then just add your
ticket number like this:
Set(CALLERID(name)=${CALLERID(name)} TICKETNUMBER:12345)
On 8/20/07, 0xception [EMAIL PROTECTED] wrote:
Is there a way, other then recoding the entire voicemail application, to
pass dialplan variables to the voicemail application and to the email
notifications of new voicemail.
For example in our small tech support queue i would like to pass the
ticket
number with the email notification that a new support voicemail was
left.
I've tried simply replacing the ${VM_WHATEVER} w/ the actual variable
name
inside the voicemail.conf file, I've also tried setting the VM variables
directly before the voicemail application call in the dial plan... both
of
these fail.
Anyone else know of another way?
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Re: [asterisk-users] Asterisk as ISDN PRI Proxy
I have done some work with Siemens hipath systems in the past - just watch out the pridialplan and it's friends. PaulH On Mon, 2007-08-20 at 16:17 +0100, Gustavo Felisberto wrote: I have a costumer with a Siemens PBX installed, and I would like to setup a Asterisk system that would act as a kind of Proxy between the Siemens PBX and the operator network. The current setup is: Siemens PBX 2*PRI - Operator what I want is: Siemens PBX 2*PRI - Asterisk BOX - Operator For the Siemens PBX the Asterisk Box would be a standard Telephony Operator, and the Asterisk box would either route the calls normally, or would route them via another system via SIP or IAX. I need to know if this is possible, and what kind of hardware do I need on the Asterisk Box to do this. I know I'll need some PRI cards to connect to the Operator, but do those cards allow me to masquerade as a Operator to the Siemens PBX? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
While hardware RAID tend to be more reliable, it is not always possible to properly monitor hardware raid in a linux system, unless you write your own code. Consider this: ~# cat /proc/mdstat Personalities : [raid1] md0 : active raid1 sdb2[2](F) sda2[1] 76139968 blocks [2/1] [_U] unused devices: none The above is from an active system that one hdd failed. It would take way longer to find such a thing on a hardware raid. Unless it came with a program that emails me notification on such a failure. On 8/20/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue Members
Thanks, that fixed it. I just looked up the bug and then patched my 1.4.10.1 source with it and it appears to work as there are now queue members listed. http://bugs.digium.com/view.php?id=10424 I can't believe nobody else ran into this. Basically the issue was that you couldn't use realtime members without having your queue in realtime queues. Now you can have a static queue with realtime members. Very useful. Peder Julian Lyndon-Smith wrote: I think that revision 80086 in the 1.4 subversion branch would fix this. Julian. Peder @ NetworkOblivion wrote: Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general] realtime_family=queue_members persistentmembers = yes autofill = yes monitor-type = MixMonitor [queue2280] music = default strategy = roundrobin timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = no joinempty = yes extconfig.conf: [settings] queue_members=mysql,ASTERISK,queue_member_table MYSQL: [EMAIL PROTECTED]:/etc/asterisk# mysql -u myuser -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 7 to server version: 5.0.24a-Debian_9ubuntu2-log Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use ASTERISK; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select * from queue_member_table; ++---+-+ | queue_name | interface | penalty | ++---+-+ | queue2280 | SIP/2224 | 1 | | queue2280 | SIP/2223 | 1 | | queue2280 | SIP/ | 2 | ++---+-+ 3 rows in set (0.00 sec) I don't see any log info for mysql, except when I manually enter the info above. I've stopped an restarted * many times. I've even tried this on two separate boxes and I get the same thing. sipeers and voicemail work, but queue members does not. Any idea? I am running 1.4.10.1. Thanks. Peder ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing Variables to Voicemail's Email Notification
Okay for a quick report back, that all seems to work...
Thanks a lot.
Not much to report back other then that :)...
On 8/20/07, C F [EMAIL PROTECTED] wrote:
After rethinking.
I'm not sure if this works, but please report back after testing.
The idea would be that the CIDNAME should not be in the subject just
the ticket number, and the ticket number should not be in the email
body just the CIDNAME.
Please try the following and report back.
exten = _X.,1,Set(BLANKS= );actual 15 spaces, since
CIDName on PSTN should never be longer, and should realy be padded
with blank spaces.
exten =
_X.,n,Set(CALLERID(name)=${CALLERID(name)}${BLANKS:${LEN(${CALLERID(name)})}})
;the above just pads the CIDNAME with blanks so you know for sure it's
at least 15 char long, yes I know if the len of cidname is longer than
blanks then blah.
exten = _X.,n,Set(CALLERID(name)=${CALLERID(name):0:15}=TicketNum:1234)
;this makes sure that it is not longer than 15 plus the ticketnumber.
exten = _X.,n,Voicemail(blah)
In voicemail.conf
emailsubject=${VM_CIDNAME:15}
If this should work then the subject should be: TicketNum:1234
emailbody=New voicemail from ${VM_CIDNAME:0:15} balh.
Again, I'm not sure this will work, please test and report back.
Thank you
On 8/20/07, C F [EMAIL PROTECTED] wrote:
While I don't have an answer on how to access channel variables from
voicemail.conf, for the problem you mention this should help.
Change CALLERID(name) to your ticket number and then use VM_CIDNAME in
the subject line.
If you don't want to lose the original CIDNAME then just add your
ticket number like this:
Set(CALLERID(name)=${CALLERID(name)} TICKETNUMBER:12345)
On 8/20/07, 0xception [EMAIL PROTECTED] wrote:
Is there a way, other then recoding the entire voicemail application,
to
pass dialplan variables to the voicemail application and to the email
notifications of new voicemail.
For example in our small tech support queue i would like to pass the
ticket
number with the email notification that a new support voicemail was
left.
I've tried simply replacing the ${VM_WHATEVER} w/ the actual variable
name
inside the voicemail.conf file, I've also tried setting the VM
variables
directly before the voicemail application call in the dial plan...
both of
these fail.
Anyone else know of another way?
___
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[asterisk-users] compatibility of PRI Two B channel transfers TBTC/2BTC
Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation. From what I've gathered, the official Digium statement is that is works with DMS100 only, and only in Asterisk 1.4.X : http://kb.digium.com/entry/26/140/ Although in a bugtracker posting with a patch from over two years ago, Matt Fredrickson from Digium says that it works with 5ESS under Asterisk 1.2.X: http://bugs.digium.com/view.php?id=3554 There are also bounties and claims of this feature working on NI2 protocol(although no patches posted) on the voip-info.org Wiki: http://www.voip-info.org/wiki/view/Asterisk+bounty+PRI+2B+channel+transfer+for+NI2+PRI+line http://www.voip-info.org/wiki/index.php?page=Asterisk%20bounty%20PRI%202B%20channel%20transfer As for actually using this feature, you apparently need to add the following lines to the zapata.conf section that you want to be able to use 2BCT: facilityenable = yes transfer=yes To execute the transfer, you need to use the Transfer cmd within Asterisk: http://voipinfo.org/wiki/view/Asterisk+cmd+Transfer And according to this post, you can only do 2BCT transfers if the first call is inbound: http://www.mail-archive.com/[EMAIL PROTECTED]/msg25131.html Does 2BCT work with DMS100 and 5ESS right now? Are there people using this in production right now that can shed some more light on exactly how they are using it, and executing the transfers? Any input would be greatly appreciated. Thanks, MATT--- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 99 bottles of beer
Steve Murphy wrote: How about this one: from an extensions.conf that someone posted on the internet, I think, and I converted to AEL; I'm sorry, but I can't find the original author. (If anybody can find his post, I'd love to give him credit.) I did test this out, and it works; just put a call to the macro ( guessgame(); ) in an extension in your dialplan Nice! While we're on the subject of silly but fun dialplan bits, check out my TV remote extension. When I moved a few months ago, there was a while when I couldn't find the wireless keyboard that I usually use as my TV remote to control MythTV. So, I built dialplan so I could use a wireless phone as my remote, instead. The dialplan reads digits from the phone and sends the correct commands to a MythTV network control interface for the frontend application. I posted my tested .conf version and the untested AEL version to the MythTV wiki. The AEL version would probably be prettier with macros, now that I think of it ... http://www.mythtv.org/wiki/index.php/Controlling_MythTV_from_any_phone_using_Asterisk -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
