[asterisk-users] Remote-Party-ID implementation on trunk
Hello, For a long tmie I am using patch 8824 in order to display the name of the called party. I do a database query and then call CONNECTEDLINE function and all works ok. I've tried the latest trunk (202183) with the same config file. It works ok for local extensions, but for extensions routed via a gateway no name is displayed. Any idea? Thanks! __Yehavi: ___ -- 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] Called party name with Cisco-2,811 gateway
Sorry for the delay... 2009/6/7 David Backeberg dbackeb...@gmail.com On Sun, Jun 7, 2009 at 4:20 AM, Yehavi Bourvineyehavi.bourv...@gmail.com wrote: Hello, I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our Nortel TX-1 PBX. Up to now I had only the calling party names passed both ways. After upgrading the Cisco to the latest release (12.4.24T) it began honoring the remote-part-ID field sent by Asterisk and sends the called name to the Nortel. However, I still do not get the called name from the Nortel to Asterisk. Has anyone managed to make this working? There are a lot of choices when you set up your interconnectivity between your Cisco gear and the T1/PRI channels. If you post your dialpeer for this I'll take a look, although strictly speaking this isn't really an asterisk question. Here is the extract from the configuratio of the gateway: voice service voip clid network-provided qsig decode allow-connections sip to sip fax protocol t38 ls-redundancy 4 hs-redundancy 2 fallback pass-through g711alaw controller E1 0/0/0 pri-group timeslots 1-31 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn overlap-receiving isdn not-end-to-end 64 isdn incoming-voice voice isdn supp-service name calling profile Network-Extension operation-value-tag local isdn negotiate-bchan isdn outgoing ie facility isdn outgoing ie extended-facility no isdn outgoing ie notify-indicator isdn outgoing display-ie isdn outgoing ie caller-number isdn outgoing ie called-number no cdp enable ! dial-peer voice 2 pots destination-pattern 0T progress_ind setup enable 3 no digit-strip direct-inward-dial port 0/0/0:15 Thanks, __Yehavi: ___ -- 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] Called party name with Cisco-2,811 gateway
Hello, I am using a Cisco 2,811 gateway to connect Asterisk over PRI to our Nortel TX-1 PBX. Up to now I had only the calling party names passed both ways. After upgrading the Cisco to the latest release (12.4.24T) it began honoring the remote-part-ID field sent by Asterisk and sends the *called*name to the Nortel. However, I still do not get the called name from the Nortel to Asterisk. Has anyone managed to make this working? Thanks! __Yehavi: ___ -- 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] RPID on SNOM phones?
Hello, I am running Asterisk 1.4 with the RPID patch (8824) in order to display the name of the *called* person. It works on Cisco and Polycom phones, but not on SNOM. The local SNOM dealer claims that he saw it working, but he cannot give me more details. Has anyone managed to make it working? Thanks, __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to avoid call waiting? Or check DIALSTATUS before Dial()?
You check for BUSY. Check for IN_USE instead. That's what I do here (on 1.4,
but I guess that 1.6 behaves similarly).
When an extension is in IN_USE state I have a decision tree after consulting
a database:
- If the user wants waiting call - dial him/her/
- If the user doesn;t want waiting call but wants voicemail answer - send
to voicemail with B prefix.
- If niether is wanted - play busy.
Regards, __Yehavi:
2009/5/15 sean darcy seandar...@gmail.com
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,ExecIf($[${DIALSTATUS} = BUSY]?Dial(${secondline},30))
But it doesn't work because * first tries Call Waiting on the main line.
Here I dial out:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2] Set(DAHDI/1-1, CALLERID=house
2127873453) in new stack
-- Executing [...@internal:3] Dial(DAHDI/1-1,
.
And now an incoming call:
-- Executing [...@incoming:1] Answer(IAX2/nhi-10929, ) in new stack
-- Executing [...@incoming:2] Dial(IAX2/nhi-10929, DAHDI/1,60) in
new stack
-- Called 1
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- DAHDI/1-2 is ringing
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- CPE supports Call Waiting Caller*ID. Sending 'Seandarcy/212 573
1432'
Is there a way to check the status of a dahdi channel _before_ dialing it?
exten =s,n,ExecIf($[DAHDI/1${DIALSTATUS} =
BUSY]?Dial(${secondline},30)) ??
What's special control 20 ??
Any help appreciated.
sean
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[asterisk-users] Polycom-330 not displaying line buddy label?
Hello, I have so far Polycom 501 and 403 which displays the label of a key and the name of a buddy near its key. Today I received new 330's and they do not display the name, only 1 2. Besides that they work correctly. Anyone has an idea, or is it a known feature? Thanks! __Yehavi: ___ -- 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] issue with sip 180 responses
Maybe it is buffering issues with the kernel? Does it happen only when there is a peak in the new calls rate? Do all 180 messages get dropped? __Yehavi: 2009/4/19 Nir Levi n...@bezeqint.co.il Hello, SIP invites are accepted from imitator , but 'SIP 180' is not responded back to imitator. By inspecting the issue , we can **see** the response is generated and sent from asterisk (via asterisk logger (sip debug )) , but while sniffing the interface with tcpdump, we can't see 180 response on the interface. We don’t have errors on the interface, firewall is disabled , seems there's no packet lost (checked with ping with low interval ) , and routes are ok … By our tests we can see there is a direct connection between the mass of the calls, and between the lost of sip 180 responses. We're using Asterisk 1.4.4, with real-time configuration, also we made few **modifications** in asterisk source code (changed app_dial.c, /app_macro.c, /func_cdr.c)… We're not sure if the problem on the OS Level (Centos 5.2) or in the asterisk application. Please assist… ~Nir -- This message was enriched by Impactia Technologies Ltd. www.impactia.com Please do not enrichhttp://impactia.bezeqint.co.il/exclude.asp?email=asterisk-us...@lists.digium.comdomain=bezeqint.co.ilid=777emails sent to me. ___ -- 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 is not designed for University with largeuser base?
Hello, After a long time we had a meeting with our university's management and got a green light to have a proof of concept with open source telephony. Now I have to select the right software to experiment with... Up to now I thought of going with OpenSER for the masses and Asterisk for voicemail and other media related things. However, from reading around it seems like FreeSwitch can give me the benefits of both packages. Anyone has an experience with it? Thanks, __Yehavi: ___ -- 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 is not designed for University with largeuser base?
Hello, Sorry for the delay - was out of office. I also cross-posting it to OpenSIPS list. I have a small pilot (20-30 phones) which also does some sort of SIP to PRI transcode for our old PBX. The pilot is base on Asterisk and mostly Polycom-501 phones. It works quite well, but I have a few minor/missing issues: - I have the RPID patch, and unattended transfers fails with it. - No SLA, only BLF. I know there is SLA, but it is cumbersome to deploy. - Confference is limited to 3 participants. I guess I can do more with external server but didn't manage yet to make it working. - No busy dial again which is required by our users. Now, to the original issue: I tried adding 1000 extensions to the SIP database, and then use SIPP to send one REGISTER for each extension. After doing so Asterisk still worked, but it was continously accessing the database for all these extensions, just polling them. This raised a red flag to me, and I decided to check the following config: OpenSIPS/Kamailo/etc. as registrar and SIP switch for the phones, while using Asterisk only for media related issues (which is the common suggestion here). Now, I have new problems: - SLA works, but very fragile. - Not BLF, although I think it will be solve with the dialog handling on OpenSIPS 1.5 - Same confference and busy dial problem. Next week our management is going to decide (I hope...) how to proceed: Do nothing (stay with the Nortel as we are tight on budget), go to open source or to a commercial solution. Although a commercial solution allows me so sleep well at night, I am going to recommend the open source direction. If accepted, then I will continue the development and you'll hear me quite a lot here asking hard questions :-) BTW, If I didn't say so far: we have around 8,000 extensions on 4 Notel PBX'es, using around 10 PRI's to the world. Regards, __Yehavi: 2009/3/17 Vincent Li vincent.mc...@gmail.com On Tue, 17 Mar 2009, Yehavi Bourvine wrote: Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: Hi Yehavi, Could you please keep us informed with your research, That would be very interesting case that all other Universities could study. There seems no known large Asterisk deployment in University enviroment at this time. 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] Asterisk is not designed for University with largeuser base?
Hello' I am at the same situation as you. I also work at a university and we have over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot. I am using a realtime users database and the main problem is that Aaterisk does too mcuh database access to inquire for the currently registered users. (I am using direct RTP path between the phones so this is not a limiting issue here). I am checking now a combination of OpenSIPS and Asterisk, where OpenSIPS will serve the phones and Asterisk the more complicate things (voicemail, transcoding, etc.). OpenSIPS still lacks some of Asterisk features, but they are being worked on. Regards, __Yehavi: 2009/3/17 Jay Milk ast-us...@skimmilk.net Danny Nicholas wrote: Sounds like a personal preference to me. Here is the Wiki for SipX. http://en.wikipedia.org/wiki/SipX Reading this, it's just another flavor of the same medicine. Both are open-source with Commercial support available. I'd contend that the business model says very little about implementation, reliability, scalability. ___ -- 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] SIP DTMF problem with SNOM
I have a similar problem with Snom. Since I've upgraded from version 6 to version 7 I cannot call IVR systems. The first DTMF goes ok, but after that others are not accepted nor I am heard by the operator at the other side. Since I am the only one who has Snom here I didn't bother to debug it... __Yehavi: ___ -- 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] Need help registering Cisco 7960 Phones on Asterisk
From my experience it won't register to the second box, only to the first one. Why? god knows... __Yehavi: 2009/1/20 D Tucny d...@tucny.com 2009/1/20 Zeeshan Zakaria zisha...@gmail.com Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXX.cnf looks like this: phone_label: Zeeshan A Zakaria line1_name: 523 line1_displayname: Zeeshan A Zakaria line1_authname: 523 line1_password: 523 line1_shortname: x523 line2_name: 523 line2_displayname: Zeeshan line2_authname: 523 line2_password: 523 line2_shortname: x523 line3_name: 224 line3_displayname: Zeeshan line3_authname: 224 line3_password: 224 line3_shortname: x224 SIPDefault.cnf contains default settings along with proxy info like this: proxy1_address: xxx.xxx.xxx.xxx proxy1_port: 5060 proxy2_address: xxx.xxx.xxx.xxx proxy2_port: 5060 proxy3_address: xxx.xxx.xxx.xxx proxy3_port: 5060 Same settings work fine from Grandstream phone, and X-lite. What am I missing on this Cisco phone configuration? That all looks fine, though I don't think the ports need quotes... You do have proxy_register: 1 too don't you? If so, check the debug output from sip set debug on, or, sip set debug peer 224 to see if it's reaching the server but not authenticating etc... Also, you can log into the phone using telnet to check status and restart registration (and many more things) d ___ -- 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] How to transfer a call from one AsteriskServerto another
Who sends the 500 failure code? Asterisk or the VOIP supplier through which you got the call? Note that Asterisk has the basic mechanism for call trasnsfer, just as you transfer a call, so the problem is either in using Transfer() inside IVR context, or the provider. As David noted - use canreinvite=yes And last word: If you get the error from the local Asterisk then raise the verbosity level - probably you'll find some hint there. Good luck, __Yehavi: 2009/1/17 David fire ddf...@gmail.com and canreinvite=yes ? 2009/1/17 Lenz Emilitri lenz.lo...@gmail.com Are you sure that the TRANSFER is supported by the other side at all? see http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267 Thanks l. 2009/1/16 Paul bulkm...@monafamily.com Yes, this is the first method I tried. The transfer only works if it is done before a media path is set up to the first box (not answered by the IVR). If it is answered then transferred, I get a 500 internal server error back from the ITSP and the call dies. I never see anything hit the second box. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 10:09 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServerto another I guess you already tried this? http://www.voip-info.org/wiki-Asterisk+cmd+Transfer Thanks l. 2009/1/16 Paul bulkm...@monafamily.com I do have it functioning with Dial(). I was looking for a way to completely move the call from the first box though. When using Dial() media moves, but the call is still tied to the first box. In looking at captures when the call is ended, the first box invites out to the ITSP again, then after receiving a 200ok sends a bye. Also while testing, once the call was up on the second box, I stopped Asterisk on the first box which kills the call. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Friday, January 16, 2009 12:17 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to transfer a call from one AsteriskServer to another Why don't you simply Dial() the call to a separate box keeping Asterisk out of the audio path? l. 2009/1/16 Paul bulkm...@monafamily.com Can anyone tell me how I can completely move an established call off of one Asterisk server to another? In our case we have a server with our IVR. Depending upon digits entered, the call can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help ___ -- 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 -- Loway - 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 -- Loway - 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 -- Loway - 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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
Re: [asterisk-users] SLA and Polycom
I also need SLA for hundreds of phones. Looking at Asterisk's way of doing it I got cold legs... I am now playing with OpenSIPS; didn't manage to make it working there, but it should be much simpler. And no, I am not telling you to drop Asterisk - it is good for other things, and even some things are done easily on it... __Yehavi: 2009/1/8 Noah Miller noahisaacmil...@gmail.com I don't believe that Polycom's version of SLA does anything with Asterisk. You have to use asterisk's SLA implementation (http://www.asterisk.org/node/48342). So asterisk can't do SLA with Polycom phones? Asterisk can do SLA with Polycom, just not using Polycom's SLA implementation (in other words, don't bother setting up shared lines in the polycom cfg files - it won't do anything). You use asterisk's SLA implementation. - Noah ___ -- 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] Good comparisons on cheaper VOIP phones
I'll have to run some TCPDUMP to see what happens. I'll also try this with OpenSIPS where there is more flexbility with the header fields. Thanks, __Yehavi: 2008/12/23 Olivier oza-4...@myamail.com 2008/12/23 Yehavi Bourvine yehavi.bourv...@gmail.com I have one ST2030 bought for testing. Indeed it has a very intuitive user's interface, bue I've found two drawbacks: - Its sound quality has some place to be improved... - It has no RPID support (displaying the name of the called party). Hi, Have you tried P-Asserted-Identity features (from latest Asterisk and ST2030) ? I 'm hoping that using this feature, you can have your ST2030 displaying the name of the person you're calling : whenever, Asterisk is acking the INVITE message it received from the ST2030, it should include the name of called party. To be honest, I really don't know how you can teach Asterisk to behave this way (ie to include called party name in ACK replies) but I'm sure the phone support displaying names from trusted parties. It's on my TODO list, anyway, as I would also like to let this phone display caller's name when picking up a call (instead of the dialed string). Regards - If these two issues are fixed, then it might be the better choice for cheaper price. __Yehavi: 2008/12/21 Olivier oza-4...@myamail.com I don't know if Thomson ST2030 SIP phones are distributed where you live but those have the best feature set-price ratio. They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...) with up to 5 simultaneous calls. Here in France, those are selected everywhere ... I would recommend them without any hesitation. ___ -- 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] Good comparisons on cheaper VOIP phones
We have one 7912 which we bought for evaluation. The main drawback is that it has hands free speaker but no microphone. __Yehavi: 2008/12/23 Mikel Lindsaar raasd...@gmail.com Thanks all for your replies. I have an aastra 9133i here for testing and am getting a polycom 320 to try out. But today, I got my hands on an older Cisco 7912G with SIP software installed. It connected fine to the Asterisk box, works with the PoE stuff I have, sounds good and doesn't seem to have any problems. Best all, I can buy near new for about $60 each in Australian dollars (thats about 45USD with the Aussie dollar being what it is :) The handsets look OK, they are nice and solid feeling and very easy to use / not complex. Any reason not to use the 7912G ? Seems with the SIP image they work just dandy... Mikel ___ -- 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] Good comparisons on cheaper VOIP phones
I have one ST2030 bought for testing. Indeed it has a very intuitive user's interface, bue I've found two drawbacks: - Its sound quality has some place to be improved... - It has no RPID support (displaying the name of the called party). If these two issues are fixed, then it might be the better choice for cheaper price. __Yehavi: 2008/12/21 Olivier oza-4...@myamail.com I don't know if Thomson ST2030 SIP phones are distributed where you live but those have the best feature set-price ratio. They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...) with up to 5 simultaneous calls. Here in France, those are selected everywhere ... I would recommend them without any hesitation. ___ -- 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] how to set the busy signal usign softphones
I use the dev_state() function to find the status of the called phone. If it is BUSY then I call the busy() application to signal a busy tone. Firthermore, I also consult a MySQL table to see whether the user wants waiting calls or not and decide accordingly. __Yehavi: 2008/12/20 nik600 nik...@gmail.com On Sat, Dec 20, 2008 at 2:25 PM, Benoit maver...@maverick.eu.org wrote: Have you tried to set the call-limit to 10 or 2 for example, i know it's what's needed for the queue system to detect busy on sip softphone Yes, but if i set the call-limit to 2 the user receive more than 1 call (correctly...up to 2 calls), even when he is busy. -- /*/ nik600 http://www.kumbe.it ___ -- 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] Country numbering plan resources
Here is the data for Israel: +972 is the international prefix, and then: 2,3,4,8,9 and then 7 digits: Landline, according to the dialling area; usually copper connected phones. 7x and then 7 digits: Landline, country wide numbering (usually IP based operators). x: 2=Golden lines (012), 3=Barak (013), 4=Globecall Partner, 6=Bezeq, 7=HOT. 5x and then 7 digits: Cellulars. x: 0=Pelephone, 2=Cellcom, 4=Orange, 7=MIRS, 9=Jawwal (Palestenian operator). 159 and then 6 digits - Country wide numbering. Used usually by small service providers who has more than one branch (like computer labs, etc.). 17xx and then 6 digits: Usually call centers; local call tariff. 18xx and then 6 digits - Toll free numbers. 19xx and then 6 digits - Premuim services. 10x - Emergency services (police, first aid, fire). 12xx - Special long term services with national importance to have short number (like psychological first aid). This is also used for temporary numbers assigned for short period (like hospital number just after some disaster). * (can also be reached with 1222) - commercial services who want short dialling string. Maybe there are more, but these are the mostly used ones. regards, __Yehavi: ___ -- 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] Large Asterisk installations (~10, 000 extensions), preferably at universities
I did a test yesterday and did 1,000 registrations to Asterisk using SIPP. I did the register test since I am using the realtime DB and asterisk does periodic quesries to it for each registered user. Although Asterisk continued to function as usuall, it was in a steady loop querying the DB for the 1,000 users. OK, you convinced me to look at some front end to it. There are mainly three front ends mentioed here: OpenSer, SipExpress and FreeSwitch. Is there some comparison available which will save me from testing all three of them? Is there one which is more used than the others? (so it has more public QA :-) Thanks! __Yehavi: 2008/11/24 Steve Totaro [EMAIL PROTECTED] Fronting with OpenSER or FS, you should have no problems providing you plan to use SIP extensions. What is critical are the max simultaneous trunks you are going to use. I would go TDM although universities have good bandwidth, and SUPERIOR bandwidth between others. I would think a TDM DS3 or two just to be safe. It should be pretty trivial besides gotchas, like cat3 to the rooms, although channel banks may be an even better solution if phones are already in place. Then you just use SIP when needed or wanted, and Asterisk is simple, although more costly. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Fri, Nov 21, 2008 at 6:24 PM, Wilton Helm [EMAIL PROTECTED] wrote: Yet another option is a commercial system with in-house staff. I used to maintain a NEC (NEAX 2400) for many years. I went to factory training and had total responsibility for it. Some manufacturers discourage or prevent this, but others are open to it. There are also 3rd party organizations (such as Source) that can supply parts and even expertise for those going that direction. Whether the result would be higher availability than Asterisk, I don't know. Given I'm both a telco guy and a computer guru (CS degree) I'd probably go the Asterisk route myself, because its open and I would have more control. Wilton and bug fixes than any commercial product sold in the intra-industrial channel ... and they won't charge you a $30,000 license fee for the upgrade. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphones with RPID and BLF
Hello, I am looking for a softphone which supports RPID (displaying the called party name) and BLF features. I couldn't find one so far... Any idea whether such a softphone exists? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Call-Waiting
Try looking at the DEV_STATE function (available separately on
Asterisk-1.4). It will tell you the status of the phone before you call the
Dial() application.
__Yehavi:
2008/11/25 Sebastian [EMAIL PROTECTED]
If is deprecated how do you treat a queue (realtime), that has to have just
one call for agent??
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: martes, 25 de noviembre de 2008 03:37 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Disabling Call-Waiting
On Tuesday 25 November 2008 10:46:49 Elliot Murdock wrote:
Thanks for the responses. I'll look into the phone devices themselves.
I am wondering, if call-limit did not solve my problem, what is the
call-limit parameter supposed to do anyway?
The call-limit is actually kind of deprecated. Using the GROUP() function
is now the preferred way to do this.
exten = 1234,1,Set(GROUP()=foo)
exten = 1234,n,GotoIf($[${GROUP_COUNT(foo)}1]?voicemail)
exten = 1234,n,DIal(SIP/tilghman,30)
exten = 1234,n(voicemail),Voicemail(1234,
${IF($[${DIALSTATUS}=BUSY]?b:u)})
--
Tilghman
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Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Thanks to everyone who replies so far! We have Nortel PBX'es with a support contract from one of the local VARs (Nortel does not give direct support here). In the last two weeks we had one of our exchanges down for three half days; one was after a failure, and the other two were when the technician came to fix remainders of the original problem and just did a 5 seconds restart which won't even cut calls. Yeh, the 5 seconds took 6-7 hours... BTW, they still do not know what was the original problem. So, why won't we save the big bucks we pay them, hire two professionals (who cost less) and support an open source code by ourselves? This way we depend on ourselves only. Thanks, __Yehavi: 2008/11/21 Grygoriy Dobrovolskyy [EMAIL PROTECTED] 2008/11/21 Yehavi Bourvine [EMAIL PROTECTED] Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: ___ -- 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 Hello very interesting project you have, however asterisk is not a registry server, i suggest that you use opensips/opense/kamalio for your registrar, from where you dispatch to you asterisk servers, inside a good environment with a controlled network and nice tagged voip flow you could acheve a good results. ___ -- 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] Large Asterisk installarions (~10, 000 extensions), preferably at universities
I know that in the past there have been people on this list who have done very large scale asterisk deployments. Not sure if any of them are still around to comment. With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). Thanks! __Yehavi: ___ -- 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] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with or without OpenSER
Hello, I am running a small installation of asterisk and looking for future expansion of it to handle thousands of users. From what I read I see that usually large installation place OpenSER (or similar solution) in front of Asterisk in order to provide high call rate because OpenSER does only signalling while Asterisk does all. My question is: If Asterisk also does only signalling (i.e. the voice traffic goes directly between the phones and not via asterisk) is it still that slow? I preffer to have one software package rather than dealing with two. Thanks! __Yehavi: ___ -- 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 with or without OpenSER
Thanks! 2008/11/18 John Todd [EMAIL PROTECTED] Hints: Will SIP redirection work for you? How about re-INVITEs? Does Linux HA give any solutions? L3 load sharing? 1. What do you mean by SP redirection in this context? 2. I am using re-invites, and made sure that the audio does not pass via the asterisk server. 3. L3 load sharing: I did some work on this and this is the method I came with: - I use MySQL with replication between two servers. - Some of the phones register to one server and the others to the other (i.e., this way I load balance). - Since the MySQL database is replicated, each server knows where the destination is registered. If it is registered on the other sevrer then the caller is redirected there. 4. HA? didn't give too much thought to it yet, but it is on my list. BTW, how do you emulate large number of clients without buying some test equipment like IXIA? Thanks! __Yehavi: ___ -- 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] One way voice after call transfer (bugs 9305, 13120)
Yes, the problem is there as well... __Yehavi: 2008/10/17 broadband Voice [EMAIL PROTECTED] Is anyone seeing it in 1.4.22? I plan to upgrade to that version, my fear is zaptel compatibility. On Fri, Oct 17, 2008 at 1:08 AM, Yehavi Bourvine [EMAIL PROTECTED] wrote: unfortunately I still see it in 1.6.0... __Yehavi: 2008/10/17 broadband Voice [EMAIL PROTECTED] I am having a similar problem and I'm using Asterisk 1.4.19 and have that problem on some calls through our calling card platforms. Someone suggested we use 1.4.3 and have not tried it yet. Any comments from the group. On Tue, Jul 29, 2008 at 1:19 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I am having an issue here that after an attended call transfer there is no audio on one way; the problem is caused by Asterisk sending two INVITE messages without waiting for an ack for the first one. The issue has been reported on bug 9305, has been fixed and the fix is now included inside the main stream (version 1.4.21). However, I still get this behaviour, so I opened a new bug (13120). This bug sits there for over a week with no reponse... Has anyone else noticed this behaviour? Any idea what I can do? My users are angry on me... Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way voice after call transfer (bugs 9305, 13120)
unfortunately I still see it in 1.6.0... __Yehavi: 2008/10/17 broadband Voice [EMAIL PROTECTED] I am having a similar problem and I'm using Asterisk 1.4.19 and have that problem on some calls through our calling card platforms. Someone suggested we use 1.4.3 and have not tried it yet. Any comments from the group. On Tue, Jul 29, 2008 at 1:19 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I am having an issue here that after an attended call transfer there is no audio on one way; the problem is caused by Asterisk sending two INVITE messages without waiting for an ack for the first one. The issue has been reported on bug 9305, has been fixed and the fix is now included inside the main stream (version 1.4.21). However, I still get this behaviour, so I opened a new bug (13120). This bug sits there for over a week with no reponse... Has anyone else noticed this behaviour? Any idea what I can do? My users are angry on me... Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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] ATA for large networks
Try AudioCodes MP-124 which is 24 ports FXS. I have one but haven't used it much yet, so I cannot comment about its quiality. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco + Asterisk
Hello, I've done similar thing with SIP. If you are interested then write me and I'll dig the details. __yehavi: 2008/9/16 Guilherme Loch Waltrick Góes [EMAIL PROTECTED] I have a Cisco 3845 with a ISDN PRI port connected to my legacy PBX, this router is running IOS 12.4(5) T5. I'm trying to integrate Asterisk with this router through H.323, I tried ooh323 (comes with asterisk-addons) and it works partially, I can make calls from Cisco to Asterisk, but the other way around dosn't work. Does anybody have any hints of what could be wrong ? -- Guilherme Loch Góes Notícias e Fórum sobre VoIP com software livre: http://www.voipexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way voice after call transfer (bugs 9305, 13120)
Hello, I am having an issue here that after an attended call transfer there is no audio on one way; the problem is caused by Asterisk sending two INVITE messages without waiting for an ack for the first one. The issue has been reported on bug 9305, has been fixed and the fix is now included inside the main stream (version 1.4.21). However, I still get this behaviour, so I opened a new bug (13120). This bug sits there for over a week with no reponse... Has anyone else noticed this behaviour? Any idea what I can do? My users are angry on me... Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changinf Polycom-501 config server from remote?
Hello, Our Polycom-501 phones are set to retreive their config for the server by a static configuation defined at the phones (boot servers). Is there any way to change it remotely? I found no relevant field in the internal WEB browser, nor anything in the configuration files (sip.conf and phone1.conf). Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] imap voicemail
Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail. I compiled c-client with the following settings: make lr5 IP6=4 and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/ However if i enable any if the imap settings in voicemail.conf, asterisk starts acting funny and dosent allow any calls imapserver=imap.gmail.com imapport=993 mapfolder=Voicemail Where did I go wrong? Note that port 993 is IMAPS and I think you are telling it to use IMAP. Note also that Asterisk can get hung once in a while when using IMAP storage. Try netstat -an | grep 5060 and see whether you have a queue building there. __Yehavi: ___ -- 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] Ring back when free?
Has anyone here implemented Ring back when free in Asterisk?
Here is what I do; the dialplan enclosed is in AEL2 format, but you can get the
idea.
When a call is originated I save the called and callee numbers in a database.
If the user gets busy he/she hangs up and dial *41. I then retreive the last
number they dialled and place a flag in the database that someone is camping on
it. The H extension checks this flag and if found generates a .call file.
First, I have a macro to save the for each extension who is the last they
called and the last who called them:
// Save the calling and called numbers in To and From and in the database so
// they can be used by *41 and *42. This way the h extension can acecss this
// database for all destinations.
macro Save_From_to ( ) {
// To and From are used in the dialplan, since we might change ${EXTEN}
Set(_To=${MACRO_EXTEN});
Set(_From=${CALLERID(num)});
NoOp(= ${From} - ${To});
// Save them in database for later use.
// Save the caller number at the called extension for *42 usage.
Set(DB(${To}/LastCaller)=${From});
// Where we called for *41
Set(DB(${From}/LastCalled)=${To});
};
This macro is called at the beginning of the normal dialplan.
Now, the *41 which registers the camp-on using the data saved above:
// *41: Camp on the last extension dialled
*41 = {
Set(tmp=${DB(${CALLERID(num)}/LastCalled)});
// Save it so when the other side hangs it will see it and dial us.
Set(DB(${tmp}/CallBack)=${CALLERID(num)});
// Say the number to caller so he can verify...
SayDigits(${tmp});
Hangup();
};
And now the H extension for handling it:
// The Hangup extension which is called when the call is hanged. See whether
// we have some waiting callback waiting on this extension.
h = {
ResetCDR(w);// To make the CDR correct.
NoOp(${From});
// We have to check the two sides of the call: Those who camp on the calling
and
// those who camp on the called.
Set(tmp=${DB(${From}/CallBack)}); // The calling.
if(${tmp} != ) {// Something is there.
DBdel(${From}/CallBack); // And delete it...
// Create the callfile and then move it to the spool directory to make the
call.
System(echo Channel: SIP/${tmp}
/tmp/test.tmp${From})
;
System(echo WaitTime: 20 /tmp/test.tmp${From});
System(echo Extension: ${From}
/tmp/test.tmp${From});
System(echo CallerID: Callback \\\${tmp}\\\
/tmp/te
st.tmp${From});
System(mv /tmp/test.tmp${From}
/var/spool/asterisk/outgo
ing/);
};
Set(tmp=${DB(${To}/CallBack)}); // The called
if(${tmp} != ) {// Something is there
DBdel(${To}/CallBack); // And delete it...
// Create the callfile and then move it to the spool directory to make the
call.
System(echo Channel: SIP/${tmp} /tmp/test.tmp${To});
System(echo WaitTime: 20 /tmp/test.tmp${To});
System(echo Extension: ${To} /tmp/test.tmp${To});
System(echo CallerID: Callback \\\${tmp}\\\
/tmp/te
st.tmp${To});
System(mv /tmp/test.tmp${To}
/var/spool/asterisk/outgoin
g/);
};
};
Good luck! __Yehavi:
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Re: [asterisk-users] BLF and Snom phones
Sorry for the hijack, but I was wondering if I could be pointed at how
to get BLF and pickup working with Polycoms? I can use their Buddy
option to get a basic BLF system working, but that does not understand
the ringing state, and does not allow call pickup.
This is what I do (it works most of the times...). Our extension numbers are 5
digits, and I'll take as example making a BLF for extension 89444:
1. In the buddies list, instead of setting the BLF to be 89444, set it to be
*789444 (i.e preppend *7 to the number).
2. change the Hint to be (I am using AEL):
hint(Sip/89444) *789444 = NoOp();
This allows the phone to send subsciptions for *789444 which is translated
to the real extension's state.
3. Define the extension *7x:
// Directed pickup by *7. When we want a line key on the phone to be both
// speed dial and direct pickup we program it to dial *7num; We strip off
// the *7 and then check the destination status; if it is ringing - pick it;
// if not - dial it.
// When picking up - Set the callerID to be the orignaly calling station.
_*7X = {
NoOp();
Set(_To=${EXTEN:-5});
Set(_From=${CALLERID(num)});
if(${DEVSTATE(SIP/${To})} == RINGING) {
Set(CALLEDID(all)=${DB(${To}/CallingID)});
Pickup([EMAIL PROTECTED]);
Hangup();
} else {
goto huji-local|_8|StartLocal;
};
}
If the destination is ringing, pick it up. If not, strip the *7 and dial it.
I am using To and From since I need them for other places.
4. At your regular dialplan set the PICKUPMARK to the extension number:
Set(_PICKUPMARK=${EXTEN});
good luck, __Yehavi:
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Re: [asterisk-users] Realtime replication!!!!!
what do you think is the best way to replicate an asterisk mysql realtime database , i'm setting up a cluster , and i obviously need to ensure the database high availability i've been reading about DRBD and the internal replication functionality of mysql , but i dont really know if its the proper solution, i'm looking for advices on how to replicate the mysql realtime database so then when the main server is down , the second replace it . I won't call it best way but I am using two-way master-slave repliaction of MySQL. __Yehavi: ___ -- 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] OT How to Change Polycom Web Admin User:Pass via Web
I am using version 2.2.0. __Yehavi: Date: Thu, 6 Mar 2008 15:01:26 +1100 From: Lee, John (Sydney) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How to Change Polycom Web Admin User:Pass As far as I recall it can be done from the config file only. Here is the relevant line from sip.cfg: device device.set=1 device.auth.localAdminPassword.set=1 device.auth.loc alAdminPassword=YOUR-PASSWORD-HERE / What sip release are you referring to? I am looking at sip 1.6.x and sip.cfg only allows you to set the length of the user and admin passwords. You cannot set the password in cfg. ___ -- 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] How to restrict a Polycom from receiving unauthorized calls
Hello, I've found that my Polycom-501 accepts INVITES from any server in the world... I would like to restrict it to accept calls only from the servers listed in its config file, but I cannot find anything in the documentation. Any idea? Thanks, __Yehavi: ___ -- 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] OT How to Change Polycom Web Admin User:Pass via Web
I could be mistaken, but I think this has to be done physically from the phone. I don't think you can do this with central provisioning or from the web interface. As far as I recall it can be done from the config file only. Here is the relevant line from sip.cfg: device device.set=1 device.auth.localAdminPassword.set=1 device.auth.loc alAdminPassword=YOUR-PASSWORD-HERE / regards, __Yehavi: ___ -- 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] load balancing
If i have this kind of setup, what do i need to make it's load balance. [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ] | | | | - | mysql cluster | - I plan on doing it via DNS SRV only, but if a user register on asterisk 1 how can users at asterisk 4 reach that user. Thank You Regards, Ron Hi Ron, If you're using realtime each Asterisk server will know where every user is irrespective of which Asterisk server they registered on. The problem is NAT, if a client is behind NAT and registers on server 1 then server's 2,3 4 are unlikely to be able to get through to it. Last time I lookedthe Asterisk realtime engine doesn't record which server an account registered on in the database so the only option I See the discussion a few days ago. The Asterisk server saves the value of SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL. Regards, __Yehavi: ___ -- 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] Load balancing SIP extensions.
Hello,
Here is how I do this. The prerequisits are:
- MySQL to hold the extensions realtime database. MySQL is synchronized
among all servers using the Master/slave replication model.
- The phones are spread by some external algorithm over the Asterisk servers
(statefull load balancer, statically defined in the config file of the
phone, etc.).
The idea is to locate on which server the destination phone is registered and
redirect the call to it. For this:
/etc/asterisk/asterisk.conf has the parameter sysname set to its IP address
(you can use also a DNS name, but I want to be independent of name
resolution). This causes the server to set the field regserver to be
saved in the MySQL database to the IP address of the server.
/etc/asterisk/extensions: The logic to check whether the value of regsever
is different from sysname and if so - redirect the call. The code
fragments are (I am using AEL):
To get the regserver from the database:
MYSQL(Query resID ${connid} SELECT regserver from sip_users where
name='${EXTEN}');
MYSQL(Fetch FetchId ${resID} RegServer);
MYSQL(Clear ${resID});
so now RegServer contains the server where the phone is registered. Next:
if((${DEVSTATE(SIP/${EXTEN})} == UNAVAILABLE) ||
(${DEVSTATE(SIP/${EXTEN})} == INVALID)) {
if(${SYSTEMNAME} != ${RegServer}) {
Transfer(SIP/[EMAIL PROTECTED]);
return;
};
};
I check for the device state so in case the phone has double registration
(primary and backup server) it will be processed localy.
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Re: [asterisk-users] load balancing SIP extensions
What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there ... I've tried doing something similar and came with two options. The common to them is that I use MySQL for realtime extensions, and set systemname parameter to the IP address of the server where the phone registers. When a call arrives I check whether the REGSERVER coloumn is the same as the local server or not. If not, then there are two options: - Pass the call via IAX to the other servers; this makes both server process the call and the audio. - Send a refer message to the caller to contact the other server. I had this working in the lab but not in production yet. If you want the dialplan code for this then email me. __Yehavi: ___ -- 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 E series Phone for SIP-VOIP calling
Hello, I've one nokia E65 that works very well with my asterisk box. The people here don't let me even try it as they are afraid it will consume the battery more than when it is used the usual way. Is this true? Thanks, __Yehavi: ___ -- 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] Cisco SIP Gateway
Is anyone using a cisco router as an ISDN gateway with Asterisk? As you might have seen from a couple of my threads, I have been looking at Fritz! and Cologne cards, both of which require development against a specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive and causes a lag in deployment. I was thinking a better approach might be to use a seperate gateway, such as a Cisco 1751 with VIC-2BRI-NT/TE talking SIP to Asterisk, much like like an SPA3K in the analogue world. Any success stories? I am using a Cisco router as a PRI gateway with SIP. I have no experience with BRI, but it should work as long as you have DSP cards for voice processing (the BRI card can do only data calls, and DSPs are needed for voice). __Yehavi: ___ -- 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] IMAP client in asterisk not trying to contact IMAP server
I'm trying to test IMAP in 1.4.17 and it appears to be not working. I've compiled imap-2007 with the following on a CentOS 5 box: make slx EXTRACFLAGS=-I/usr/include/openssl -fPIC and I've configured and compiled asterisk with the following: ./configure --with-imap=/usr/local/src/imap-2007 And now in make menuselect you have to go to voicemail options and set IMAP support to on. Here's my voicemail.conf: [general] imapserver=localhost imapfolder=Inbox ;pollmailboxes=yes ;pollfreq=30 imapflags=notls authuser=asttest expungeonhangup=yes authpassword=whatever I had to enable pollmailboxes in order to update MWI. __Yehavi: ___ -- 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] disable call waiting by default
I've connected some analogic phone to some fxs modules on an analogic card. I want to disable by default the call waiting sound. I know that dialing *70 before to call the call waiting is disabled until the next call, but isn't there a setting or a dialplan command to set up this automatically? If you mean that there is no waiting call then use DEV_STATE function to see whether the called extension is in a call; if so - call the Busy() application. If you need more details then search in the lists's history - I;ve posted a code fragment about a month ago. __yehavi: ___ --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] b2bua
On Friday 04 January 2008 16:45:00 ameel wrote: Is there a way to disable the b2bua feature in asterisk. I would like asterisk to work as a sip server and not be involved in the RTP path between phones. No. And by the way, b2bua is not a feature. It's is literally what Asterisk is. If I understand correctly the question the RTP can be established directly between the phones and does not have to pass through the Asterisk PBX. Just set CANREINVITE=YES and use the same CODECs for both phones. __Yehavi: ___ --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] enable eyeBeam to accept only one call
Hello
I'm using eyeBeam, and Asterisk keeps sending my clients a second call,
when they are still in one call (because eyeBeam has lots of channels).
I was using X-Lite (with 3 channels) and Asterisk never sent the client
a second call.
How can I force Asterisk (or eyeBeam) just to send one call at each time.
Is this a configuration I need to do in eyeBeam or Asterisk?
Please contact Eyebeam and request this feature - The more people who
ask for it the better. I have requested it previously to no avail.
It is possible to emulate this in asterisk using inuse counting, but I
still get the occasional off-by-one that requires an asterisk restart
to bring a phone back to life.
I am doing that on 1.4.15 and it works fine so far. The code fragment is:
if(${DEVSTATE(SIP/${To})} != NOT_INUSE) {
Busy();
};
Note that for function DEVSTATE you have to install the relevant function. See
http://www.asterisk.org/node/48360
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Re: [asterisk-users] retrieve last number dialled
What is the easiest (simplest) way to do this?
I do it in two steps: Save the dialled number in Asterisk DB and have a
special extension (*41) which redials it.
Here is the abstract from the dialplan where I save it:
Set(_To=${EXTEN}); // Save the original extension dialled.
Set(_From=${CALLERID(num)}); // Save the caller.
Set(DB(${To}/LastCaller)=${From});
Set(DB(${From}/LastCalled)=${To});
and then the special extension:
*41 = {
Set(tmp=${DB(${CALLERID(num)}/LastCalled)});
SayDigits(${tmp});
// Save it so when the other side hangs it will see it and dial us.
Dial(SIP/${tmp});
};
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[asterisk-users] Finding the status of an extension
Hello, I would like to check whether an extension is busy or not before calling the Dial() application to it (for example - to play a Busy if it is on conversation). How do I check it? In the trunk version there was a function DEVSTATE(SIP/123), however it does not exist on version 1.4.13... What is the equivalent of it? Thanks, __Yehavi: ___ --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] Need T1 crossover cable?
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? As people answered here you need a crossed cable; Note that T1/E1 cables are different than Ethernet. T1/E1 uses pins 1,2,4,5 (while Ethernet uses 1,2,3,6). A crossed T1 cable is: 1 4 2 5 4 1 5 2 Regards, __Yehavi: ___ --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] Polycom phones and corporate phone directory
Hello, A few days ago I've posted two questions about Polycom phones: How to access corporate phone directory from the phone and how to use a conference server with it. After I got zero responses I tried openning a support call in Polycom's site. Here are the replies I got from them: - Corporate directory: They are thinking about it, probably will use LDAP. I was asked to open a feature request to which I got no response yet. - conference server: I asked it because there is a a parameter to set an external conference server, and when I tried it only the initiator was transfered to the server; the other parties were left in hold state. Polycom replied that it is not supported and only the phone's internal conference feature should be used. Regards, __Yehavi: ___ --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] Unlocking Cisco 7921
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For the purpose of the pilot (i.e. low investment) I want to configure the phones from the keypad. Each phone shows settings locked! whenever I try to edit the network profiles. I can't seem to unlock them! Hopefully there is a secret button combination...I would hate to have to go to a Cisco Unified CallManager just to unlock a few phones... On other Cisco modemls the sequence to unlock them (as long as you did not set a password) is **# (star, star, pound sign). I hope it works on them as well. __Yehavi: ___ --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] IMAP usage with Asterisk
Yehavi Bourvine +972-8-9489444 wrote: In any case, I'll try this week to upgrade to 1.4.6 version and then add IMAP support and inform what happens. There have been _many_ IMAP related fixes sine 1.4.6. Please try the latest version, 1.4.13, instead. -- Russell Bryant Sorry, I had a typing mistake - I meant 1.4.13... Anyway, I tried it at the lab today and it works ok. There is one minor problem when parsing the IMAP headers (files a bug at ID 11043). I'll be able to move it to the production system only next week. Thanks, __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions about Plycom phones and Asterisk
Hello, I have two issues which I would like to know whether someone has an answer to them: 1. Our institute has over 8,000 phone numbers and I would like to allow people to search it from the phone. I am willing to write some XHTML scripts to run through the microbrowser, but I cannot find at the documentation what format it should be in order for the phone to understand the output and allow dialling of the numbers returned. 2. Polycom supports external conference server via the parameter voIpProt.SIP.conference.address. Any idea how should I include in it the conference room number (so I can pass it to MeetMe() application)? Thanks! __Yehavi: ___ --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 and Cisco
Isee no one has answered you so far, so I'll try: I have asterisk ip-pbx on my network, with some grandstream ip phone and i have cisco gateway that is connetced to VOIP service providers . Cisco is 3700 series and is using H323 . i have compiled H323 on asterisk . now i want to make a call from ip phone that is registerd to asterisk , and route call to VOIP provider so this call should goes to teh cisco from asterisk and then go to the VOIP serviece provider network ip-phone -asterisk-cisco-voip netwrok how can i do that , whats asterisk config file ( H.323, sip and extensions.conf ) and how is cisco configuration for this setup? Thanks Giti I've done this with Cisco-2,811 (the exact model should not matter) and with SIP (H.323 should be quite similar). On Asterisk you have to add a rule in the dialplan to direct the outgoing calls to the Cisco. In the most simple way it should be something like: Dial([EMAIL PROTECTED]) On the Cisco gateway you should have a dial-peer which calls the PSTN. For example: dial-peer voice 2 pots destination-pattern 0T progress_ind setup enable 3 no digit-strip direct-inward-dial port 0/0/0:15 (this will direct any call to 0xxx to voice card 0/0/0:15). If you need a more detailed config please contact me off-list. Regards, __Yehavi: ___ --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] IMAP usage with Asterisk
Hello, I tried a few months ago to use IMAP with Asterisk; I used either 1.4 or the latest SVN at that time (sorry, don't remember). After a day I had to remove it since Asterisk crashed, mostly in the IMAP client code (the code of UW IMAP). My users wants IMAP back (they loved it) but not in the price of crash... I could not reproduce the crashes at the lab. They only occour on the live system with users. I suggest to write the IMAP client code by the Asterisk developers and not depend on external code. In any case, I'll try this week to upgrade to 1.4.6 version and then add IMAP support and inform what happens. Thanks! __Yehavi: ___ --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] Difference between trunk and released versions
Hello, Up to a while ago I thought that the released versions are checkpoints of the trunk versions; however, now I understand they are not, as I see differences between the two trains. So, what is the relation between them? Examples for differences: - When the language is different than Engligh the trunk version is reading numbers from /var/lib/asterisk/sounds/Lang-Name/digits while the release version is using /var/lib/asterisk/sounds/digits/Lang-Name - MAILBOX_EXISTS function is replaced with MailboxExists application. - External IVR has no way to exit from the program under the release version... The documentation is correct with the trunk version. Thanks! __Yehavi: ___ --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] Difference between trunk and released versions
Hello Sean, Does this clear things up? Yes! Thanks! __Yehavi: ___ --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] extensions.conf vs. AEL
Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --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] online active call watching
try the astman command. __Yehavi: ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Show Callee name on Display
SInce no one else has brought this up, just thought I'd let you know that it is being worked on... http://bugs.digium.com/view.php?id=8824 And it works - I am using it for months already. Note that not all phones support it. Cisco and Policom supports it, while Snom does not. __Yehavi: ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] ExternalIVR() broken in 1.4.9?
Hello, After upgrading to 1.4.9 the above function does not work anymore; it claims that child went away while the child is probably not born at all... Before I open a bug on it, anyone has a clue? Thanks! __Yehavi: ___ --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] Looking for a USB phone handset or headset
Hello, I would like people to use soft phone, but they are used to have the standard phone handset in their hands... Is there a USB handset or a handset that connects to the audio card? Thanks! __Yehavi: ___ --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] TCP-UDP SIP proxy?
Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SLA with SIP-only environment
Hello, All the examples of SLA talk about Zap channels from one side and SIP on the other side, while my system is a pure SIP one. I would like to have two phones having extensions 1 2 defined on them, and when someone calls extension 1 it rings on both, each one can see its status, and when one station puts line 1 on hold the other one can pick it. Is it possible at all? If so, can someone give the relevant fragments for sla.conf, sip.conf and extensions.conf? Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Call waiting tone
Hello,
A few days ago I've asked about the ability to play a stuttered ringing
tone when the called party is already on the phone. I've found a partial
solution for it.
To describe again the problem: When a user is on a call and someone else
calls him, the caller does not know that the called party is on the phone
(while the called party wants to know that someone else is calling him/her and
not just play busy). On our public PSTN the caller is notified by a stuttered
ringing tone (thus he can decide whether to wait or hangup and call later).
What's I've done is that when the called party is on the phone a short message
is sent to the caller (may be also a recording of the stuttered ringing) and
then the call is passed. Here is the code fragment:
exten = _806XX,n,Set(Status=${DEVSTATE(SIP/${EXTEN})}) ; Get his status
exten = _806XX,n,GotoIf($[${Status} == NOT_INUSE]?OK:WAITING_CALL)
exten = _806XX,n(WAITING_CALL),Playback(waiting-call)
exten = _806XX,n(OK),Dial(SIP/${EXTEN}${aEXTEN},20,) ; Dial the phone for 20
Hope it helps someone...
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[asterisk-users] Call waiting tone when calling a busy station?
Hello, When dialling a SIP phone which is already in a call the caller hears a regular ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call waiting tone when calling a busy station?
this is a SIP phone configuration issue. You should tell the UAC to not accept a second call while the line is engaged (look for a 'Call Waiting' option in the configuration of the UAC) The UAC will send back a 486 Busy Here error code and the calling party will get a busy signal from asterisk This is not what I meant. I want the called party to get a sign of a waiting call and answer it if he/she wants. I want the caller to know that he on a waiting call (here it is customary to play a stuttered ring tone). in short - can I signal in the 183 ringing packet that this is a second call? Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Called party identification - where to take called name?
Hello, I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Called party identification - where to takecalledname?
Yehavi wrote: I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? Asnwering myself: I am using realtime extensions, so I've added call to MYSQL() application to get the called user callerid field. __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MYSQL application in dial plan
Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open? Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] How to find out line that you are on from Bezeq
Does anyone know the code to dial in Israel on Bezeq to get back the number of the line that you are on ? Dial 110 and it will tell you your number. __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SLA with SIP only configuration
Hello, I would like to understand how SLA is working, but all the examples are using trunk of ZAP type and stations of SIP. However, in my case the stations are SIP and the outgoing connection to PSTN is SIP also (link to a Cisco gateway). can anyone send me a simple configuration of how to make SLA between two SIP phones without dependency of anything else? It seems I lack some basic understanding of thios feature and maybe such an example can help. Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connection between Asterisk - Cisco 2851
Hello from Greece. I have an assignment related to Cisco. Specifically, i have to connect an asterisk server (VoIP) to a Cisco Router 2851 in a way that the router will handle some calls. I have not found any manuals that would be helpful to me. On the other hand, i found some manuals for connection between asterisk and Cisco Call Managers. I would like to know if the call manager is a part of the router or is an extra element. I would appriciate if you could help me. I'll just add one more thing: Your Cisco must have DSPs in order to allow the E1 to have voice calls (on 2,811 it is called PVDM). Without the DSPs your E1 can do only data calls... __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco GW, PRI CallerID Name
Does anybody have callerid name coming in on a Cisco PRI via a Cisco gateway via SIP to *? I've seen a few people ask and a few people that say it should work, but I've never seen an actual working config. I have it working, but it depends on the specific configuration. I have it working via PRI/Q.sig to a Nortel TX-1. The key issue here is that the name can be sent in two ways: via the Q.931 and via Q.sig. Nortel supports only Q.sig, and you need to enable it on the Cisco. If this is your case then tell me and I'll send the config. __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - Automatic Redial on No Answer
Here are the relevant parts from extensions.conf file. It works only for local
extensions whose number id 806xx. Note one thing: When you use the H extension
the generated CDR is wrong - the destination extension is H and not the
original number. I've done some small code change in Asterisk and has to file a
bug about it.
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exists,
; send busy if no mailbox. Same for busy.
; We try to avoid the n+101 rule whenever possible, but it is not always
; possible as HasVoiceMailbox() does only n+101 jump.
;;; First, save the caller and called numbers for *41 and *42 features.
exten = _806XX,1,Set(_To=${EXTEN}) ; Save the original extension dialled.
exten = _806XX,n,Set(_From=${CALLERID(num)}) ; Save the caller.
; Save the caller number at the called extension for *42 usage.
exten = _806XX,n,Set(DB(${To}/LastCaller)=${From})
; Where we called for *41
exten = _806XX,n,Set(DB(${From}/LastCalled)=${To})
; See whether this extension needs to ring at others as well. To define such
; an extension do inside the command prompt of Asterisk:
; database set EXT-NUMBER MoreLinesToRing SIP/firt-lineSIP/Second-line
exten = _806XX,n,Set(aEXTEN=${DB(${EXTEN}/MoreLinesToRing)})
; Now dial the extension.
exten = _806XX,n,Dial(SIP/${EXTEN}${aEXTEN},20,) ; Dial the phone for 20
seconds.
; No answer or busy
exten = _806XX,n,GoTo(s-${DIALSTATUS},1) ; Jump according to the failure
mode
exten = _806XX,n,Hangup() ; Just to be sure...
; No answer:
exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-NOANSWER,n,Busy(); No maibox = play busy.
exten = s-NOANSWER,102,VoiceMail(${To}|u) ; Has mailbox - send the call to
there
; Busy:
exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-BUSY,n,Busy(); No maibox = play busy.
exten = s-BUSY,102,VoiceMail(${To}|b) ; Has mailbox - send the call to there
; Congestion: Handle the same a busy.
exten = s-CONGESTION,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-CONGESTION,n,Busy() ; No maibox = play busy.
exten = s-CONGESTION,102,VoiceMail(${To}|b) ; Has mailbox - send the call to
there
; Unavailable channel - act as busy:
exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1);
; Called here when the call is successfull and the user hanged the phone.
; Check whether the user has a waiting callback queued on him/her
exten = h,1,NoOp(${From} ${To} ${EXTEN})
exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for us
exten = h,3,NoOp(${From} ${tmp})
exten = h,4,GotoIf($[ ${tmp} ]?5:103) ; Anyone waiting for us?
exten = h,5,DBdel(${From}/CallBack); And delete it...
; Create the callfile and then move it to the spool directory to make the call.
exten = h,6,System(echo Channel: SIP/${tmp} /tmp/test.tmp${To})
exten = h,7,System(echo WaitTime: 20 /tmp/test.tmp${To})
exten = h,8,System(echo Extension: ${From} /tmp/test.tmp${To})
exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\
/tmp/test.tmp${To})
exten = h,10,System(mv /tmp/test.tmp${To} /var/spool/asterisk/outgoing/)
; No one is waiting for us...
exten = h,103,NoOp(Nothing to call)
; The activation codes for the above code.
; *42: Get the last number who called us, say it and call it.
exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller})
exten = *42,n,SayDigits(${tmp})
exten = *42,n,Goto(${tmp},1)
; *41: Camp on the last extension dialled
exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)})
exten = *41,n,SayDigits(${tmp})
; Save it so when the other side hangs it will see it and dial us.
exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)})
exten = *41,n,Hangup()
Good luck, __Yehavi:
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Re: [asterisk-users] SIP - Automatic Redial on No Answer
Hello, I've wrote a dialplan script which uses the H extension to do something similar to what you want. In general it uses the internal ASTDB for this: - When there is no answer (or busy) the caller hangs up, initiate a new call with some special code (*41 is used here by the public carrier, so I am using it also). Asterisk registers the data in its DB. - When the user disconnects the H extension is called. It then looks in ASTDB to see whether there is a user camoing on this extension. If so, a call file is created and Asterisk initiates the call. If this is what you need please tell me and I'll post the code on Thursday. I've already posted it in the past so you might search the archives in the meantime. __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference server (or how to make a call with more than 3 u
On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: Hello, On most SIP phones a conference call is done on the phone and is limited to 3 participants. Polycom phones has a configuration option to use a conference server instead of the internal conferencing feature. I guess I need some conference server; any experience with such a server which can interact with Asterisk? Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Conference server (or how to make a call withmore than 3 u
Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. I have a few Snom phones here - people do not like them... Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference server (or how to make a call with more than 3 u
Yehavi, Can you make a script that uses call files to get everyone into the conference? -- Warm Regards, Lee Possible, but looks too much cumbersome... However, that's a nice idea. Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conference server (or how to make a call with more than 3 users)
Hello, On most SIP phones a conference call is done on the phone and is limited to 3 participants. Polycom phones has a configuration option to use a conference server instead of the internal conferencing feature. I guess I need some conference server; any experience with such a server which can interact with Asterisk? Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco PRI gateway config
That was exactly what I meant. Your setup is : Nortel --- Cisco --- Asterisk What I was thinking about is: Nortel --- Asterisk1 --Asterisk2 In previous case, your are using Cisco's QSIG features. In the latter one, you could use Asterisk QSIG features. I was asking because, I was wondering how Cisco and Asterisk QSIG features compare. Sorry, I have no idea... __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SetCIDNum is not available on 1.4svn
Hello, I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it on 1.4svn 56126 and it does not recognise this application. Any idea?... Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco PRI gateway config
do you have some info, about what platforms supports qsig decode? I found, that first supported in 12.4.(9)T, but I don't know, if only on 28xx or also in older routers (like 3660) with NM-HDV-E1... I assume that it is a hardware independent but I do not have the hardware to test... I see refernce to this command in 12.3T documentation, so it should be inside the normal 12.4 version. I moved to 12.4T due to TAC request before I tried the above command, so I did not test it on 12.4. __yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco PRI gateway config
Do you think it could have been done with another T1/E1Asterisk box between the Nortel PBX and the other Asterisk server ? Sorry, I do not understand exactly what you are asking. Do you mean using an Asterisk with PRI card instead of Cisco? If so, I have no experience with this. Which features would you then loose or gain, given current status of QSIG support in Asterisk ? In my case the Cisco did all the Q.sig work so Asterisk's Q.sig functionality was not used. __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Cisco PRI gateway config
Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined to send and receive names via Q.sig. The definition fragments on Cisco are: isdn switch-type primary-qsig (so it will use Q.sig signalling). ... voice service voip qsig decode(This sends names out via Q.sig) fax protocol pass-through g711alaw sip controller E1 0/0/0 pri-group timeslots 1-31 ... interface Serial0/0/0:15 (This is for E1 PRI). no ip address encapsulation hdlc isdn switch-type primary-qsig isdn overlap-receiving isdn not-end-to-end 64 isdn incoming-voice voice isdn supp-service name calling (This receives names via Q.sig) isdn negotiate-bchan isdn outgoing ie facility isdn outgoing ie caller-number isdn outgoing ie called-number no cdp enable Anc the rest is quite standard. Regards, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco PRI gateway config
interesting! so it means, that you can now see caller id names between sip phones connected to asterisk and phones connected to pbx? Yes! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lastest SVN (1.4) and realtime call limit
Hello, I am running version 1.4 with realtime support. I've set (for Snom phones 300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the database). - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem was that when such a phone received a call and did attended transfer it was left in use and could not receive new calls. - After seeing reference to similar problem on this list I;ve downloaded today the latest SVN source code and installed it. The problem is that it shows the call limit as 0 and not as 1. Any idea? Thanks, __yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7912 phones loosing registration
I had a similar problem with Asterisk-1.4; After digging slightly I've found that sterisk crashed (safe_asterisk restarted it and I did not notice) and this caused the phones to lose registrations until the next expiration... I am not sure this is your case, but try TCPDUMP of port 5060 from these phones. It might give a clue. __Yehavi: Date: Thu, 15 Feb 2007 13:57:47 -0500 From: Jerry Geis [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] 7912 phones loosing registration I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router for supply a connection from PABX to Asterisk
anyone know if they have a solution in Cisco for: 1- Connect old PABX (with BRI or PRI) to a cisco router 2- Connect this cisco router in SIP to a Asterisk Server I am search if cisco can this and what is the modele for this I am using a Cisco to connect Asterisk via PRI to our Nortel TX-1. The Cisco is a voice bundle of 2,811 + E1 + PDLM card. Note that you need PDLMs as the same number as the PRI channels you are going to define (i.e. 32 PDLMs for each PRI). I am controlling the Cisco via SIP; it works, but a few problems: - Only basic connectivity. No additional features (like names) as the Cisco supports them only via MGCP (in MGCP is passes all the Q.sig signals to the PBX - Asterisk in this case - and it should do all the handling, but I did not find how to do it with Asterisk). - Cisco has no authentication mechanism, so anyone which has access to port 5060 of it can generate calls. Asterisk is not better when it concerned in such sutiations... Regards, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM phones stay in use after transfer
Hello, Before I dig into the SIP traces maybe someone has a clue: We have a few Snom phones, and the call-limit on the extension associated with them is set to 1 (so the other Snom phones can blink the relevant LED). When I call a SNOM phone (300 with the latest firmware) and the call is answered the phone is in in use state (verbose printing from Asterisk). When I hit the transfer button it gets into hold state in Asterisk. I dial the new extension and the call seems to be transferred (actually it goes to hell...). Anyway, the Snom is now in left in hold state forever, and Asterisk won't accept new calls to it since it has max calls of 1... Similar thing happens with attended transfer (putting the call on hold, calling the other party and either disconnecting or pressing Transfer). Any idea? Thanks! __Yehavi: BTW, any idea when 1.4.1 is scheduled to be released? I need the SLA feature desperately... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco PRI gateway with MGCP control
Hello, Anyone managed to control a Cisco voice gateway (2,811 in my case) using MGCP? I cannot make the PRI going on-line (while with SIP I can). If you ask why I want to use MGCP and not SIP: it is because Cisco uses different Q.sig signalling when you manage it with different protocols, and I need the other Q.sig... Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Cisco 7960 functionality
The users do not want to push the new call softkey or the speaker button in order to dial a call. They want to be able to just begin dialing the number. The users do not want to push the answer softkey after they pickup the handset in order to answer a call. Doesn;t it answers when you pick up the handset? Here it does so... The users want the transfer softkey on the screen while on a call. Currently it is acessable via the more softkey. I've asked Cisco whether all the above can be done and got a negative reply. __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI/Q.sig between Cisco Nortel
Hello, I am using a Cisco-2,811 as a gateway between the Asterisk PBX and our Nortel TX-1 university's PBX. It is working but no names are exchanged. From the debug mode I see that the Cisco sends the display name (which does not appear on the Nortel's phones) and the Nortel does not bother to send it at all. I recall that when I had a pilot with Cisco CCM two years ago we had to set the siganlling to ESGF on the Nortel and use MGCP on Cisco (since MGCP protocol forces the Cisco to use ESGF signalling). We could not use H.323 as it forces the Cisco to use ISGF. I suspect that SIP is the same, but setting ISGF signalling on Nortel doesn't help. Anyone had some luck with this configuration? Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback/ringback
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exists,
; send busy if no mailbox. Same for busy.
; We try to avoid the n+101 rule whenever possible, but it is not always
; possible as HasVoiceMailbox() does only n+101 jump.
exten = _999XX,1,Set(_To=${EXTEN}) ; Save the original extension dialled.
exten = _999XX,n,Set(_From=${CALLERID(num)}) ; Save the caller.
; Save the caller number at the called extension for *42 usage.
exten = _999XX,n,Set(DB(${To}/LastCaller)=${From})
; Where we called for *41
exten = _999XX,n,Set(DB(${From}/LastCalled)=${To})
; Now dial the extension.
exten = _999XX,n,Dial(SIP/${EXTEN},20,) ; Dial the phone for 20 seconds.
; No answer or busy
exten = _999XX,n,GoTo(s-${DIALSTATUS},1) ; Jump according to the failure
mode
exten = _999XX,n,Hangup() ; Just to be sure...
; No answer:
exten = s-NOANSWER,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-NOANSWER,n,Busy(); No maibox = play busy.
exten = s-NOANSWER,102,VoiceMail(u${To}) ; Has mailbox - send the call to there
; Busy:
exten = s-BUSY,1,MailboxExists(${To}|j); Has a mailbox?
exten = s-BUSY,n,Busy(); No maibox = play busy.
exten = s-BUSY,102,VoiceMail(b${To}) ; Has mailbox - send the call to there
; Unavailable channel - act as busy:
exten = s-CHANUNAVAIL,1,Goto(s-BUSY,1);
; Called here when the call is successfull and the user hanged the phone.
; Check whether the user has a waiting callback queued on him/her
exten = h,1,NoOp(${From} ${To} ${EXTEN})
exten = h,2,Set(tmp=${DB(${From}/CallBack)}) ; Get who is waiting for us
exten = h,3,NoOp(${From} ${tmp})
exten = h,4,GotoIf($[ ${tmp} ]?5:103) ; Anyone waiting for us?
exten = h,5,DBdel(${From}/CallBack); And delete it...
; Create the callfile and then move it to the spool directory to make the call.
exten = h,6,System(echo Channel: SIP/${tmp} /tmp/test.tmp${To})
exten = h,7,System(echo WaitTime: 20 /tmp/test.tmp${To})
exten = h,8,System(echo Extension: ${From} /tmp/test.tmp${To})
exten = h,9,System(echo CallerID: Callback \\\${tmp}\\\
/tmp/test.tmp${To})
exten = h,10,System(mv /tmp/test.tmp${To} /var/spool/asterisk/outgoing/)
exten = h,103,NoOp(Nothing to call)
; *42: Get the last number who called us, say it and call it.
exten = *42,1,Set(tmp=${DB(${CALLERID(num)}/LastCaller})
exten = *42,n,SayDigits(${tmp})
exten = *42,n,Goto(${tmp},1)
; *41: Camp on the last extension dialled
exten = *41,1,Set(tmp=${DB(${CALLERID(num)}/LastCalled)})
exten = *41,n,SayDigits(${tmp})
; Save it so when the other side hangs it will see it and dial us.
exten = *41,n,Set(DB(${tmp}/CallBack)=${CALLERID(num)})
exten = *41,n,Hangup()
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[asterisk-users] Disallowing unauthorized calls to Cisco Polycom phones
Hello, I would like the IP phones to not accespt SIP requests (like INVITE) from any device other than its proxy. Snom phones ignore this while Cisco Polycom accepts the call. Any idea what to do to disable it? Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] real life example of SLA definition
Hello,
I am looking for a real life example of using SLA lines under Asterisk.
I'll describe my environment and would like to know how I define it in
Asterisk (version 1.4 final).
Suppose I have two multi lines phones. The first phone has extension 1
assigned to it, and the second phone has extension 2 assigned to it. Now, I
want extension 3 to be available on both phones as additional shared extension.
What shall I define on the phone? (define extension 3 on both doesn't work as
only one can register with it). What should sla.conf file have? Do I have to
change extensions.conf? (To make it simple let's assume that it contains only
Dial(SIP/${EXTEN}) as the dialplan).
Thanks! __Yehavi:
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Re: [asterisk-users] real life example of SLA definition
You say you can't put 'extension 3' on both phones because only one can register to it, but that is incorrect. Extensions and SIP devices are not the the same thing; extensions exist in the dialplan, and SIP devices are in the SIP channel driver. Create two SIP friends for the two phones to use for their 'extension 3' line appearance (call them 'phone1-ext3' and 'phone2-ext3' or something), get them both to register, and then dial both devices from the dialplan when extension 3 is called. OK, but how do I make that when extension 3 is being used on one phone the other one can see its status, and when it is placed on hold then it can be picked-up on the other phone by just hitting the line button? I am trying to emulate the current phone system behaviour, as this is what the users want... Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
