[asterisk-users] RDNIS question
Perhaps I've got the whole concept wrong, but here goes:
Using 1.4, when someone from the outside dials my direct line (123456),
I want it to call my extension at work (SIP/456), my extension in my
home office (vpn connection to corporate lan, SIP/678) and my mobile
(654321). So my dialplan is thus:
exten = 123456,1,Dial(SIP/456SIP/678Zap/G3c/07803654321,30)
exten = 123456,n(VMNoAnswer),Voicemail(${EXTEN}|su)
exten = 123456,n,Hangup()
this works well, with one exception: when I take the call on the mobile,
the callerid info is the number of my switchboard. I presume that this
is because I am dialling out from the switch board.
Enter RDNIS. I added an extra line to the dialplan
exten = 123456,1,Set(CALLERID(rdnis)=${CALLERID(number)})
exten = 123456,n,Dial(SIP/456SIP/678Zap/G3c/07803654321,30)
exten = 123456,n(VMNoAnswer),Voicemail(${EXTEN}|su)
exten = 123456,n,Hangup()
However, I get exactly the same result (callerid of the switchboard, not
the original caller). And yes, I did a reload ;)
Any help would be appreciated.
Julian
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Re: [asterisk-users] using a mobile phone as a handset via bluetooth
Am Samstag, den 09.12.2006, 14:37 +1100 schrieb James Harper: Normally when you think of using Bluetooth with mobile phones you think of using it to attach a headset wirelessly to a mobile phone... can it work the other way? Can I have a Bluetooth card on my laptop/desktop such that my mobile phone can be a handset to a softphone on the laptop/desktop? Yes, but... For Bluetooth devices, the appropriate gremium defined profiles, specifying capabilities for certain device types and how those shall interact. Most mobile phones seem to implement the phone profile allowing to connect a headset profile device. Obviously those are different specs! And I did not yet find a mobile phone that implements the headset profile correctly. A possible alternative could be running a full blown SIP client on those phones that support it. There is SIP client software for at least Windows CE, I do not know wether Palm, Java phones or whatever will work. Of course this is only reasonable if your phone supports wireless 802.11 or another means of over-the-air networking (leaving bluetooth, GPRS is not interesting for SIP transport). A pity that Windows CE seems to NOT support Bluetooth TCP/IP networking, at least in those two incarnations that I ran into. It seems mobile network providers do not like their customers to have an alternative choice besides GPRS data transfer :-/ I would love to hear if there are phones that actually implement the headset part. Regards, Anselm ___ --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] 5.8gig phone MWI
Tom Lynn wrote: You're trying to teach a pig to sing. The uniden items you refer to probably have their own internal answering machine, mine does. It's designed to light the lamp only when it's own machine has a message. You're giving out totally incorrect information. The TRU-8866 unit I mentioned is a 2 line unit (which I wanted), but does NOT have a built in answering machine (which I didn't want). Uniden seems to offer models with and without answering machine function. However, despite the fact that it does not have a built in answering machine, the handsets and base unit both support MWI. I believe that Uniden does make a single line version of this phone, and I bet it also supports MWI - especially since they've standardized their handsets to be universal across their latest line. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel module compile woes
Hi all, I'm pretty new to linux and compiling modules, but I've scoured the web for help on compiling the zaptel modules from source and I get the following error... make -C SUBDIRS=/usr/src/modules/zaptel modules make: *** SUBDIRS=/usr/src/modules/zaptel: No such file or directory. Stop. make: *** [linux26] Error 2 Any ideas? /usr/src/modules/zaptel is the dir I'm running make linux26 in. This is a Debian server and I've retrieved the zaptel-source via apt-get. Any help greatly appreciated! Best regards, Phil ___ --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] 5.8gig phone MWI
Well that is not exactly true. I nelieve it was clear as Steve stated below that his Uniden MWI does work with Asterisk. Many phones advertise compatibility with phone company MWI. Often people use phone company VM and then have stuttered dialtone as well as FSK signaling to tell the user there is a message waiting. There are definitely phones that will do this I am just trying to fine out which ones do and actually work! Asterisk through many FXS's would send this same signal. Doug On Fri, 8 Dec 2006, Tom Lynn wrote: You're trying to teach a pig to sing. The uniden items you refer to probably have their own internal answering machine, mine does. It's designed to light the lamp only when it's own machine has a message. On 12/8/06, Doug Crompton [EMAIL PROTECTED] wrote: Thanks, but unfortunately that is an expensive 2 line phone compared to others in their line that have a base and two or three remotes for the same price. Seems a lot to pay for a MWI. I wonder if anyone has had experience with panasonic wireless 5.8gig and MWI?? They advertise compatibility on some models but I also saw a review comment that it did not work. Doug On Fri, 8 Dec 2006, Steve Prior wrote: Doug Crompton wrote: Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug I've tested the MWI with the Uniden TRU-8866 phone and it works for me. I've tested it with the Digium TDM400P FXS. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel module compile woes
On Sat, Dec 09, 2006 at 12:07:28PM -0500, Phil Finkler wrote: Hi all, I'm pretty new to linux and compiling modules, but I've scoured the web for help on compiling the zaptel modules from source and I get the following error... make -C SUBDIRS=/usr/src/modules/zaptel modules make: *** SUBDIRS=/usr/src/modules/zaptel: No such file or directory. Stop. make: *** [linux26] Error 2 Any ideas? /usr/src/modules/zaptel is the dir I'm running make linux26 in. This is a Debian server and I've retrieved the zaptel-source via apt-get. Any help greatly appreciated! Which Debian? generally you should simply use: m-a a-i zaptel Also: run: apt-get install kernel-headers-`uname -r` -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Quality Metrics
I've been reading about ${RTPAUDIOQOS}, which supposedly contains call
quality metrics, but I can't seem to convince Asterisk to grab it for me
after the call.
Anybody have any luck with it?
Anybody know how to get Asterisk to executive a command after the call is
over?
Thanks,
Eric
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Re: [asterisk-users] Running Asterisk on a Home rotuer
- Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, December 08, 2006 12:07 PM Subject: Re: [asterisk-users] Running Asterisk on a Home rotuer Dovid B wrote: tacking pn = adding on - sorry for not being more specific. I have seen that people in the past have used a linksys router to run asterisk. It would be to expensive to bring in a PC for every location. So we want to import cheap home routers put asterisk on them as use them as the go in between the IP phones and the asterisk server. Check with Brian Capouch. He deployed Asterisk on Linksys WRT54G in some rural areas. Caveat here: Cheap = not enough horses :). Don't expect to pass many calls through one of those things. You might want to look at deploying a lightweight SIP proxy on the router instead of asterisk. Leo Ping Brian Capouch. Anyone have his contact info ? ___ --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] Realtime question
- Original Message - From: Rob Schall [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 05, 2006 7:14 PM Subject: [asterisk-users] Realtime question Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any suggestions? Thanks. Rob If you dont turst a single mysql machine than how about a cluster ? ___ --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] Quicknet PhoneJack questions.
Hi,
i have already bought this card and successfully configured IXJ driver in
kernel but i have few problems:
1. I have no dialtone, somtimes it appears for a very small time and dissapears.
I have in phone.conf configured mode=dialtone
2. Second progress inband, when number is placed i hear ringing tone even if
called party is busy, my voip operator passess progress inband properly - with
softphone connected to my asterisk im able to hear ringing/busy/invalid number
etc.
3. The last thing is problem with dialplan when i set:
exten = _00.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:2}
after press only 00 the system tries to callout, not waiting for other digits.
Please help if you have experience in PhoneJACK PCI.
Regards,
Adam Rybak
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[asterisk-users] H.323 trunk
Hi all, How to configure H.323 trunk to H.323 provider from Asterisk System? How can i check trunk status? Where i put your user account and password on the ooh323.conf file? UGA ___ --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] oh323.conf question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, December 07, 2006 5:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] oh323.conf question Hi all, I would like to know if it exists the possibility to send to different context according to the caller IP Addres I receive H323 calls, and I have to route this to different devices according to the caller ip. I tried to use the context=first-context alias=99 context=second-context alias=88 but I was not able to succed in this; Moreover, I think the keyword alias is related to the phone calling more than to the ip address, and it could be anything... In other terms, what I have to do is to send all the calls from one IP Address to a zap group, and all the calls coming from another IP to another zap group. Any help will be gratly appreciated, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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 __ NOD32 1912 (20061209) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
Hello List, The main issue is server selection regarding PCI bus connectivity for Asterisk solutions. Most offerings on HP Proliants, something I've been looking into, include PCI-X and/or PCI-e expansion slots. I know PCI-e is totally different from PCI and PCI-X so, for now, that's not an issue. However, regarding PCI and PCI-X, and after googling for a while and checking wikipedia and whatnot, I'm still not clear on my main issue: Can one use a PCI interface card in a PCI-X slot ? If so, under what conditions ? (ex: 3.3v cards only ? PCI-X bus speed is brought down ? what ?) The objective would be to use Digium's echo cancelling PRI and BRI cards and/or beroNet's BRI cards on, for example, a Proliant ML350 G4 or G5 containing PCI-X slots -- would such combination be technically feasible ? If not, where are you guys getting servers for your PCI based solutions ? Can anyone shed some light into my doubts ? Pointers, documentation, experiences ? The second, kind of attatched issue, is associated to the growing PCI-e buses in the current servers. I know Sangoma already has a PRI card for PCI-e. What about Digium ? Other PRI, BRI manufacturers ? Thanks in advance and regards to all, -- Ex Vito ___ --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] 5.8gig phone MWI
The ad at this Panasonic site is what is so confusing. They mention a MWI on the phone, which has a digital answering machine as part of the system. Then in the text they mention the MWI requires phone company subscription to voice mail. http://www2.panasonic.com/webapp/wcs/stores/servlet/vModelDetail?storeId=15001catalogId=13401itemId=96903catGroupId=25039modelNo=KX-TG5671S Doug On Sat, 9 Dec 2006, Steve Prior wrote: Tom Lynn wrote: You're trying to teach a pig to sing. The uniden items you refer to probably have their own internal answering machine, mine does. It's designed to light the lamp only when it's own machine has a message. You're giving out totally incorrect information. The TRU-8866 unit I mentioned is a 2 line unit (which I wanted), but does NOT have a built in answering machine (which I didn't want). Uniden seems to offer models with and without answering machine function. However, despite the fact that it does not have a built in answering machine, the handsets and base unit both support MWI. I believe that Uniden does make a single line version of this phone, and I bet it also supports MWI - especially since they've standardized their handsets to be universal across their latest line. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Realtime question
-Original Message- From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Saturday, December 09, 2006 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime question - Original Message - From: Rob Schall [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 05, 2006 7:14 PM Subject: [asterisk-users] Realtime question Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any suggestions? Thanks. Rob If you dont turst a single mysql machine than how about a cluster ? You could also try realtime static. If your MysQL db crashes, Asterisk will continue to operatate until a reload. ___ --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] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
Ex Vitorino wrote: Hello List, The main issue is server selection regarding PCI bus connectivity for Asterisk solutions. Most offerings on HP Proliants, something I've been looking into, include PCI-X and/or PCI-e expansion slots. I know PCI-e is totally different from PCI and PCI-X so, for now, that's not an issue. However, regarding PCI and PCI-X, and after googling for a while and checking wikipedia and whatnot, I'm still not clear on my main issue: Can one use a PCI interface card in a PCI-X slot ? If so, under what conditions ? (ex: 3.3v cards only ? PCI-X bus speed is brought down ? what ?) The objective would be to use Digium's echo cancelling PRI and BRI cards and/or beroNet's BRI cards on, for example, a Proliant ML350 G4 or G5 containing PCI-X slots -- would such combination be technically feasible ? If not, where are you guys getting servers for your PCI based solutions ? Can anyone shed some light into my doubts ? Pointers, documentation, experiences ? The second, kind of attatched issue, is associated to the growing PCI-e buses in the current servers. I know Sangoma already has a PRI card for PCI-e. What about Digium ? Other PRI, BRI manufacturers ? Thanks in advance and regards to all, -- Ex Vito ___ --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 The rule of thumb is if it fits you can use it unless it doesn't work, there are few that won't (Creative's soundcards being an example of ones that don't) -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
Andrew D Kirch wrote: The rule of thumb is if it fits you can use it unless it doesn't work, there are few that won't (Creative's soundcards being an example of ones that don't) I remember reading up on it and (other than there being 2 different types of PCI-66 slots and then there's the PCI-100 and PCI-X ones), discovering that a particular PCI card should work in a PCI slot. When I put the card in I blew a £300 server board. Oops. ___ --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
good Linux references (was: Re: [asterisk-users] Re: Switching from FreeBSD to Linux - which distro?)
On Friday 08 December 2006 14:03, John Novack wrote: David Thomas wrote: Note: a default install of CentOS installs a bunch of unnecessary services that you will want to turn off using chkconfig service_name off. David It MIGHT be useful for SOMEONE to specify what those unnecessary services are John Novack That depends entirely on what services you need to have running. No one will hold being inexperienced against you- but unwilling to learn is something else. The world is crammed to the gills with excellent Linux system and network administration references. A good resource for anyone new to CentOS is the Red Hat manuals: http://www.redhat.com/docs/ And of course as all good computer geeks know, go to the distribution's home page, which is http://www.centos.org/. Wikis, forums, mailing lists are all here. Running a complex server like Asterisk has several interdependent parts: Asterisk itself, provisioning phones, operating system administration, and network administration. You already know about this here excellent list, and hopefully are aware of http://www.voip-info.org/wiki/, the home of all things Asterisk. These are my fave resources for Linux system and network administration: TCP/IP Network Administration, Third Edition http://www.oreilly.com/catalog/tcp3/ TCP/IP is fundamental to VoIP and computer networking. If you don't understand TCP/IP everything else will remain mysterious bash Quick Reference http://www.oreilly.com/catalog/bashqr/ get up to speed quickly on the primary Linux command shell Open Sources: Voices from the Open Source Revolution http://www.oreilly.com/catalog/opensources/book/toc.html Free to read online. My favorite inspirational book with essays by the movers and shakers of the free software movement And of course, may I modestly point to my own book (see sig.) Though it's getting a bit old and needing an update. Carla -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power 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] Running Asterisk on a Home rotuer
Dovid B wrote: - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, December 08, 2006 12:07 PM Subject: Re: [asterisk-users] Running Asterisk on a Home rotuer Dovid B wrote: tacking pn = adding on - sorry for not being more specific. I have seen that people in the past have used a linksys router to run asterisk. It would be to expensive to bring in a PC for every location. So we want to import cheap home routers put asterisk on them as use them as the go in between the IP phones and the asterisk server. Check with Brian Capouch. He deployed Asterisk on Linksys WRT54G in some rural areas. Caveat here: Cheap = not enough horses :). Don't expect to pass many calls through one of those things. You might want to look at deploying a lightweight SIP proxy on the router instead of asterisk. Leo Ping Brian Capouch. Anyone have his contact info ? See his post to the dev list. Not sure if the address is still valid. http://lists.digium.com/pipermail/asterisk-dev/2004-December/008181.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Jabber Client
Hello, I would like to connect jabber client (Exodus) to Asterisk 1.4. Asterisk 1.4 already supported jabber (client/component), didn't it? Can I connect Exodus to Asterisk directly? And if yes, please tell me how to connect? Kawamoto ___ --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] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
...well, thanks Andrew + Thomas, but that's exactly what I am trying to avoid: knowing by trying ! ;-) I can't risk spending a few thousand just to reach the conclusion that Digium's PRI or BRI cards do not work with a particular system's PCI-X slots/bus... Or, worse, staying with a dead card / system board in my hands ! :-( Anyone ? Thanks + regards, -- Ex Vito On 12/10/06, Thomas Kenyon [EMAIL PROTECTED] wrote: Andrew D Kirch wrote: The rule of thumb is if it fits you can use it unless it doesn't work, there are few that won't (Creative's soundcards being an example of ones that don't) I remember reading up on it and (other than there being 2 different types of PCI-66 slots and then there's the PCI-100 and PCI-X ones), discovering that a particular PCI card should work in a PCI slot. When I put the card in I blew a £300 server board. Oops. ___ ___ --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] Anonymous clid ?
Hi for put a anonymous clid on a out line sip, what is the config ? thanks bye ___ --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 forgetting about client registration or Polycom phone forgetting to register?
While what you say might/should help, it doesn't fix the problem. Additional information, since posting this question, till now, everything worked fine, since it wasn't a working day and only 3 manager sessions are open to asterisk, I'm suspecting that it has to do either if there are lots of phone calls going on, or when the manager has more than 3 active connections. On 12/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote: I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. ... host=dynamic We've found much better results with the static IP here. Can you try this? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --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] RDNIS question
Julian Lyndon-Smith wrote:
snip
this works well, with one exception: when I take the call on the
mobile, the callerid info is the number of my switchboard. I presume
that this is because I am dialling out from the switch board.
Enter RDNIS. I added an extra line to the dialplan
snip
2 issues here:
a. For PSTN, you should use Set(${CALLERID(num)}) to set your outgoing
caller id.
b. Does your PSTN line allow you to set the outgoing caller id? If
you're using analog, it's not possible. For ISDN (both BRI/PRI), it's
usually possible if you subscribed to the feature. But, you're normally
only allowed to set the caller ID to one of the numbers allocated to
your ISDN line. You can't just set it to any arbitrary number (Note:
might work if your local exchange is mis-configured).
Leo
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Re: [asterisk-users] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
I can't risk spending a few thousand just to reach the conclusion that Digium's PRI or BRI cards do not work with a particular system's PCI-X slots/bus... Or, worse, staying with a dead card / system board in my hands ! :-( Anyone ? I don't know about Digium cards, but I just installed a Sangoma A101 card into an IBM server in a PCI-X slot and it is working perfectly. You should ask Digium hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
