Re: [asterisk-users] Busy Lamp Fields
On Fri, 2010-07-16 at 17:34 +0100, Paddy Grice wrote: Seems BLF only work on called extensions - is there a way to show busy for the calling extension? You don't say what version of asterisk you are running this on, or have any config snippets, so difficult to say what might be wrong. Check that in the [general] section of sip.conf, you have: limitonpeers=yes Also try setting for each extn in sip.conf, call-limit to some value. I have the following: call-limit=4 In each sip.conf extn entry. (In this case I don't think the actual number is that critical, just that it must be set to something, otherwise the BLF keys don't seem to work properly.) This may only apply to asterisk versions prior to 1.4 though. Then do a reload from CLI. Rob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 Asterisk server, multiple registrations to ITSP
On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote: Hello list ! Having troubles with multiple registrations to one and the same ITSP. This sip.conf : register = user1:pass...@sip.itsp register = user2:pass...@sip.itsp ; outgoing conversations [user1-out] type=peer host=sip.ITSP Try setting type=friend instead of peer for these and see what happens. -- Robert Lister - email/sip: r...@lentil.org - http://www.lentil.org tel: 020 7043 7996 -- _ -- 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] is roundrobin and rrmemory the same meaning?
On Wed, 2010-01-13 at 10:42 +0800, Zhang Shukun wrote: is there some function used to login a agent automaticlly like agentlogin(agentname,agentpassword,phonenumber)? Depends what version you are running. AgentCallBackLogin() is deprecated and you should not use it. But the feature can be reproduced with dialplan logic. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd% 20AgentCallbackLogin This is a whole world of pain, as is using Agents in some situations. It is better to use SIP channels. (Agents do not seem to work nicely with a bunch of other features.) It is less flexible. It may be better for you to do this using AddQueueMember and RemoveQueueMember on SIP channels, and program a key (or keys) on the handset to add and remove the member from the queue dynamically instead of adding them as static members in queues.conf. Rob -- _ -- 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] Beginners Guide to setting up a Call Centre
Do you have any idea of numbers of users, and number/type of external lines as this can be quite important when deciding what type of asterisk setup and hardware to go with. (For example, if your lines are already presented over ISDN PRI or BRI, or if they are provided over IP, by an IP telephony provider.) Also you will need to think if you want to support analogue devices such as modems/fax machines etc. Do you have existing IP handsets that you want to integrate, and what are these? Or are you starting from scratch? Or are you going to use PCs with soft phones and headsets? (Often very suitable for a call centre setup) What sort of support do you require for the system / handsets? Do you need CTI integration / soft phones / headsets etc? How many lines in total are coming in to the system? Do you need hotdesk users or are they all based at the same desks every day? What are the requirements for redundancy/failover? (ranging from 'none' to 'magic failover between two sites') If you can answer this, then it will help work out what sort of hardware you will need (software can be changed about to suit, but choice of server setup/cards/media gateways is important in that decision as well.) Software, There are many pre-built solutions that are based on asterisk which have GUIs to use/admin them. These may or may not do what you want out of the box. Hot desk support is particularly limited in many of these. Or you can install just the base asterisk and roll your own. This is a bit more complex (and maybe unneeded if you are using on the most common features.) but it has its benefits, such as not being restricted by a particular GUI or management system, and being able to customise things a bit more. Rob On Tue, 2010-01-12 at 10:55 +, Peter Childs wrote: This is currently still at a proof of concept stage. After being mis-sold a Alcatel phone system, that does None of the things we asked for (Ok it takes calls but that's about it) We are looking at alternatives to try and bring some of the features we previously had on our old Analogue STS phone system. Looking at all the docs I can find Asterisks looks like it should be able to do the job and a whole lot more. This is for a small call centre so ideally we want all the features of an average call centre, ACD, Call Recording, Queue's etc etc. Any pointers on how to get started would be most helpful. -- _ -- 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] Question about SIP registration
On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote: Then I have configured an account as following: [999] type=friend username=999 You don't appear to have a secret= line in there with a password option... or did you snip it? Can someone explain me this kind of behaviour? Is it normal? Can I restrict registration of 999 peer only to SIP UA from network 1.1.1.X? There is an ACL option for the SIP peer which you can add, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip +permit-deny-mask (although there were some issues with this in earlier versions of asterisk.. it should work properly in recent versions.) Rob -- _ -- 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] is roundrobin and rrmemory the same meaning?
On Tue, 2010-01-12 at 11:26 +0800, Zhang Shukun wrote: Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring another? roundrobin is deprecated in 1.4 and you probably shouldn't use it, but rrmemory is probably what you want, trying each extension in order, but continuing the position in the queue where it left off for subsequent calls. roundrobin always starts at the top of the queue and works along rrmemory remembers which queue member was tried last, and continues for subsequent calls from where it left off, rather than starting again from the top of the queue. In 1.6, the old roundrobin behaviour (or equivalent) is renamed linear and rrmemory is renamed roundrobin If you want to add some dialplan actions for queue members, have a look at PauseQueueMember and UnpauseQueueMember which allows for queue members to be 'in' and 'out' of the group (although if using Agents then you will probably want to implement agents logging in and out), but you could replace agents with dynamic queues and program buttons on the phones which dial codes to pause and unpause the queue member. Rob -- _ -- 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] Attempted break in ?
On Mon, 2010-01-11 at 10:45 +, --[ UxBoD ]-- wrote: Hi, I am starting to see a lot of these: [Jan 10 01:18:56] NOTICE[5627] chan_sip.c: Call from '' to extension '33155786056' rejected because extension not found. [Jan 10 01:52:47] NOTICE[5627] chan_sip.c: Call from '' to extension '033155786056' rejected because extension not found. [Jan 10 02:26:36] NOTICE[5627] chan_sip.c: Call from '' to extension '0#33155786056' rejected because extension not found. Yes, looks like it. Make sure that your sip.conf context= default context points to a context that cannot make external calls. (Or, if your asterisk box does not need to accept connections from anyone externally then restrict what can connect to it with firewall rules or an access-list.) Although I had locked down the SIP config already, I was almost caught out recently by one of these attackers, where somebody was trying to make calls over *H323* as that ALSO has a 'default' context similar to sip.conf (although the calls did not succeed because before an outbound call is placed, we check the caller ID is within an expected range, in order to set the correct outbound CLI, but were that check not in place, then it probably would have succeeded.) H323 seemed to be enabled by default, so I just disabled the H.323 module as we do not use it. Rob -- _ -- 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] Questions about Voicemail
On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote: Regarding the email to multiple receipients, it is available on an ad-hoc basis from the phone? IE; call into the voicemail system, enter x digit to send a voicemail to multiple users, record the message, then enter the destination mailboxes, separated by # sign... You can enable an option in the voicemail that allows the prompt: 'To send a message to another user'... sendvoicemail=yes ; Allow the user to compose and send a voicemail ; while inside VoiceMailMain() [option 5 from ; mailbox's advanced menu]. If set to 'no', ; option 5 will not be listed. This would enable the option from within the vm app, but you want to do a dynamic list of mailboxes to deliver to, so by the time we get here, I think it's going to be to late to to anything useful (since we already called the voicemail app.) You could write some dialplan magic with a while loop, so that the user can dial a specific extn (maybe call it 'group message') and then it will prompt for a mailbox number, followed by #, or just # to end. Then it could build this list of mailboxes as a variable before calling the voicemail app. I can attempt to build an example if you are interested. Rob ___ -- 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 many lines do you use.
On Wed, 2009-11-25 at 08:46 -0500, David Gibbons wrote: I use two ‘lines’ though ‘Line appearances’ would be a better term, though still confusing in my book. One line for incoming, one line that auto-answers for paging. Cisco really has so many line appearances on their phones to enable BLF using SIP over TCP. Cisco 7960 does not do BLF (at least not on the SIP firmware) but the 7961 might. It's a shame they haven't added such features, but there we go.) If you enable two line keys with the same user/pass then the phone will automatically put a second call/call waiting onto the second line key (assuming you have call waiting enabled.) But personally I preferred the way it presented the second call before, on a single line, and found the way it displays it with two lines a bit confusing. (I can't remember exactly why now, something like it would flash the second line icon but not show you the call information until that key was pressed, or you scrolled to it.) I could see users not getting on with this, so I didn't configure it like that. The rest can be used for speed dials, but these were of limited use to me since for some reason, although the line keys can be provisioned remotely over TFTP, the speed dials cannot. It's okay for personal use though. Personally moved off my 7960 in favour of the SNOM 370 as this supports far more features than the Cisco SIP image, which is only really a piece of migration fluff to enable Cisco to migrate customers away from competitors SIP systems onto Call Manager with the dual-boot/application loader. The SNOM perhaps doesn't look as fancy as the Cisco handset, but it wins hands-down on SIP features. (the remote provisioning system was a little complicated to set up, but once set up it's okay.) It's a shame since the Cisco is a very capable (and expensive) handset, just let down by no development in the software other than small bug fixes for many years. ___ -- 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] Questions about Voicemail
On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote: I am sorry if this is not the correct place to post a question. I am wondering if there is way in asterisk 1.2 to: 1. Send a voicemail (from the phone) to multiple recipients. Yes I believe so. 1. The voicemail app allows delivery to multiple destinations at once: - example : exten = 100,1,VoiceMail(u101102103) 2. Create an e-mail alias/list and deliver the voicemail via e-mail to that alias. 2. Require (as an admin) for users 1st logging into their voicemail to change their greeting and/or password. There is a user option forcegreetings: forcegreetings = [yes|no] Sets whether the user will be forced to record a new greeting when logging in to the system for the first time. Default: no Example: forcegreetings = no Not sure about the forced change PIN, but it should be easy enough to write a little command wrapper around it and prompt for PIN via the dialplan. Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
On Wed, Dec 03, 2008 at 11:33:08AM -0500, Mike wrote: Hi, Been playing with Call parking, and I can`t help but wonder if I am doing something incorrectly. The way I understand it (using default config in features.conf), is I would transfer a call to extension 700, which would park the call, tell me 701. I could then hang up, go fetch the fright person and tell him call 701 you have a call waiting for you. The way I have it working now, is that I can transfert the call to 700, I do get 701 as a response but then, the call doesn't leave my phone. The caller gets put on hold (great) but I also get put on hold and need to keep the call going. If I hang up, so does the caller. Can`t the parked call just go park itself (and hang up my leg of the call), and ideally call me back if not picked up within x seconds? You need to complete the attended transfer when you hear the holding music, and then the call will go away from your phone. Then dial the park number i.e. 701 to get the caller back. It sounds like maybe your handset is conferencing or joining the two calls together somehow rather than doing an attended transfer. I would check how you do attended transfers normally and if you get the same symtoms. If that's okay, then it might be a failure to negotiate the right codecs etc when asterisk tries to complete the transfer, so worth checking for any no compatible codecs errors on the console and checking the codec the handset is using is one that's supported and configured for that client in sip.conf (For example you might be able to transfer between two handsets configured to use G.729, but asterisk is probably using G711 ulaw / alaw or GSM codec, for the call park channel, so this might explain it not working.) Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
On Wed, Dec 03, 2008 at 10:56:48AM -0600, Danny Nicholas wrote: The way I made this work was to set up 200 as my parker and I do transfer, 200, transfer. exten = 200,1,Answer exten = 200,n,Park(701) That will work but only for one call park slot. If that's what you want then great. If you have multiple users then surely you'd need some way to find a free slot first? (Or maybe just allocate every extension its own unique parking slot, but they'd only be able to park one call at a time?) Rob ___ -- 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 for transfer
On Tue, Dec 02, 2008 at 05:04:25PM +0530, Max Alex wrote:
Hi All,
I need to stop the transfer feature on particular sip user.
I am using linksys phone and it has set the forwarding enable to another
user.
I have three users 2101, 2102, 2103.
2102 is registered in linksys phone with forwarding enable to 2103.
But is there any procedure in asterisk that we can not allow 2102 not to
forward on 2103.
and also i want to prevent the SIP/2.0 302 Moved Temporarily.
please advice me that how can we set the user for not to forward or transfer
on 2103.
i have tested with allowtransfer=no in sip.
I'm a bit confused as to what you are asking as you mention two things,
call forwarding, and transfer.
I take this to mean call forwarding (aka divert) is where one handset
is set up to divert its incoming calls to another handset.
Transfer is where the user has answered the call, and then wants to
transfer an active call to another extension.
To make matters more interesting, there are multiple ways to do both
forwarding and transfer, depending on your configuration. Both can be done
either by the handset, or by asterisk doing it in-band (if you so configure
it in features.conf, and asterisk is actually in the media path.)
You don't mention what model Linksys phone you are using, but it may be
possible to disable the Call forward features, or to lock it so that the
number can't be set or changed by the user.
That might be sufficient, or you may want the user to be able to forward
their phone to other numbers, just not 2103. This might be slightly harder
to achieve, but you might be able to arrange that the SIP accounts are in
different contexts in sip.conf / extensions.conf such that the context that
2102 is in, does not include 2103, or just responds with Congestion() or
some error tone when 2103 is called, and so 2102 won't be able to actually
dial 2103, and hence the divert/transfer won't work either.
If 2102 actually needs to be able to dial 2102, or you have some other call
group problems, then maybe what you want is to have your dialplan calls
come in to Queue or Agent or Local device instead of a SIP device, then you
may have a bit more control over it in the dialplan. (You can't transfer
calls to an Agent.) But this might break too much other functionality you
want to keep (call waiting etc.)
Another approach would be to disable the call forward features on all the
handsets and put in some dialplan logic that uses astdb or some other source
to process call diverts.
This is what I do, so it would be fairly trivial to put in some dialplan
logic in extensions.conf, if I wanted to, to prevent certain users from
being able to divert to certain other users. (Users have to use a little web
interface or phone interface to set up their divert destinations on the
server.)
I'm not sure how you could easily prevent a call transfer from the handset
for one specific destination though.
If you were really desperate you could disable the handset's local transfer
features (if the handset allows that) and do all transfers with asterisk
(#1 / #2 etc.) Then you could control by using the callerID and
${TRANSFER_CONTEXT} what transfers to where. This would mean, however, that
asterisk would have to stay in the media path for all calls to do with that
handset, which could make it slightly less efficient in some setups, (say
for example the server is remote to the handset in a different site. A call
between two handsets even in the same site, would have the media path going
via asterisk in another site.)
You might be able to monitor the number of concurrent SIP channels the
handset can have (maybe see GROUP() variable or various other sip options
for limiting maximum number of channels) Then have some logic that does not
allow the second extension to be called if the first extension has 1 or more
calls in progress (the assumption being that the first extension is trying
to transfer to the second one.)
Anyone else think of some nicer ways to do this?
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Re: [asterisk-users] Call parking
On Wed, Dec 03, 2008 at 11:13:49AM -0600, Danny Nicholas wrote: This actually works for multiple slots. When 701 is occupied, * finds next defined slow. Does it announce what that slot is before doing it? Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 134-138 Borough High Street, London SE1 1LB Registered in England 3137929 at 3 Park Road, Peterborough, PE1 2UX ___ -- 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] canreinvite=yes problem
On Wed, Dec 03, 2008 at 06:23:32PM +0100, BERGANZ François wrote: Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outbridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... I believe canreinvite=yes is the default option unless you set it to canreinvite=no I would leave it set to yes unless there is some reason to change it, for example the phone is behind NAT, or transfers etc don't work correctly without it being set to no. If it's still not doing the right thing, then it's worth also checking the nat= option There are also other settings which can cause asterisk to stay in the media path, as BOTH sip devices need canreinvite=yes, otherwise it will stay in the media path. Specifying certain options on the Dial() cmd may also cause it to stay in the media path. Rob ___ -- 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] Parking calls
On Wed, Dec 03, 2008 at 03:27:28PM -0200, Sebastian wrote: The thing is I have to wait checking a database value to change the state, that duration is not long, but on any case I don't know when will be ready to go on. If I use MusicOnHold app the dialplan get stuck there and there's no further movement on my dialplan lines. I will have a while loop checking for a database value to change, if it changes the call will go on through the dialplan depending on the result, but I can't make the call wait without any sound (I thought PlayTones could be a possibility but I prefere MOH). For these reasons I can't use a shell script launched in background. Is there any way to launch in background some app like Background but follow with the next dialplan line while it plays the sound?? (Just like Ringing does on my solution), I know making a local channel is not the best solution, but at this moment I can't think on a different one that not involves agi. In passing, you may be interested in: http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels I haven't tested but it seems to suggest a possibility for Using the local channel to play music on hold when already answered, while waiting for a script that takes a while. ___ -- 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] twice normal beep before busy tone ??
On Fri, Oct 31, 2008 at 08:18:32AM +0100, Stefan Guenther wrote:
Hi,
I have a strange problem with our Asterisk installation. Outgoing calls
are handled by the following lines:
exten = _0[2-9]X.,1,Set(CALLERID(num)=0403${CALLERID(num)})
exten = _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} =
040321]?04030:${CALLERID(num)})})
exten = _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr)
exten = _0[2-9]X.,4,GOTO(fehler,s-${DIALSTATUS},1)
What happens if you do Answer() before the Dial?
Rob
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Re: [asterisk-users] Blank Voicemail.Conf after Password Change
On Wed, Oct 29, 2008 at 01:13:56PM -0400, Leah Newmark wrote: From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707 2008-10-29 12:14 /etc/asterisk/voicemail.conf I believe what it does it create a new file called voicemail.conf.new in the same directory and then copies it into place, so worth checking the permissions on the directory as well, that asterisk can write to it. Asterisk is running as asterisk: 24560 ?Ssl 409:34 /usr/sbin/asterisk -U asterisk I see your asterisk is running -U asterisk but this ps output is ambiguous. What does ps xaguwww show? if it really is running as UID asterisk, you should see something like: asterisk 8506 0.0 0.6 443672 12912 ? Ssl Oct02 31:46 /usr/sbin/asterisk -U asterisk -G asterisk Nothing generated from voicemail is showing up in the asterisk logs, nor does the console show any error after changing a password. Otherwise, it could be some sort of odd file locking issue where multiple things are trying to write to the same file at once? Or perhaps you have a blank voicemail.conf.new that it can't erase, sitting about somewhere? Maybe try running asterisk under strace to see what happens when you try to change a password. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 134-138 Borough High Street, London SE1 1LB Registered in England 3137929 at 3 Park Road, Peterborough, PE1 2UX ___ -- 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] twice normal beep before busy tone ??
On Fri, Oct 31, 2008 at 11:39:31PM +, Robert Lister wrote:
On Fri, Oct 31, 2008 at 08:18:32AM +0100, Stefan Guenther wrote:
Hi,
I have a strange problem with our Asterisk installation. Outgoing calls
are handled by the following lines:
exten = _0[2-9]X.,1,Set(CALLERID(num)=0403${CALLERID(num)})
exten = _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} =
040321]?04030:${CALLERID(num)})})
exten = _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr)
exten = _0[2-9]X.,4,GOTO(fehler,s-${DIALSTATUS},1)
What happens if you do Answer() before the Dial?
Also try without the r option to the dial command:
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Rob
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[asterisk-users] Asterisk voicemail message order
Hello, Anyone know if there is a way to reverse the message order for saved voicemail messages in asterisk (1.2.x)? For example, when I listen to a new message and it moves to the Old folder, the next time I retrieve messages from Old, start with the most recent message rather than having to press 6 lots of times to plough through 20 messages to get to the most recent message? (Or, an option to skip to the last message in a particular folder?) Regards, Rob ___ -- 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] Documentation of users.conf
On Mon, Sep 01, 2008 at 12:11:31PM -0500, Nestor A. Diaz wrote: Hello, does anybody know where is documented every parameter of the users.conf file in the asterisk distribucion tarball ? I believe that this is the same format as sip.conf and it's included from sip.conf in asterisknow setups, but it has a mix of settings from the other files. Entries that you define manually should probably be in some other file (sip.conf etc?) Rob ___ -- 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 7960 audible hold reminder?
On Fri, Aug 15, 2008 at 12:27:16PM -0500, [EMAIL PROTECTED] wrote: Hello, I have recently setup my first PBX and am wondering if there might be a way to send audible notification to the cisco 7960 phone when a call is put on hold. We lost a call due to a customer being on hold and forgotten about (yikes). Is there a way to get the phone to beep or ring down the same or other SIP channels after a certain amount of time on hold? Yes and no. (I am on the SIP version 8.9) In the config file for the phone: call_hold_ringback: 1 This option means that if there is a call on hold, and the handset is replaced (say, after ending another call) then the held call will ring again at the handset. I don't think there is a way (on the handset) to set a held call timeout to re-ring on the phone. If you park the call with asterisk instead of holding it, then the call park option allows calls to come back to the person who parked them after a set timeout. You may be able to do something else in asterisk, though not sure what. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 134-138 Borough High Street, London SE1 1LB Registered in England 3137929 at 3 Park Road, Peterborough, PE1 2UX ___ -- 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] UK GMT/BST settings
On Wed, Mar 26, 2008 at 10:43:13AM -, Adrian Marsh wrote: Ah ok, Those settings do seem to work (test phone was going to a different tftpd server..) Anyone know if the Ciscos re-download SIPDefault.cnf periodically, or only on boot ? As far as I can see, only on reboot. You will need to send the phones a SIP notify to get them to reboot. (or go round and reboot them all) from asterisk: (sip notify cisco-check-cfg ) Where sip_notify.conf contains: [cisco-check-cfg] Event=check-sync Content-Length=0 Give it about 20 seconds after sending the notify and the phone should reboot. R. -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ -- 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] Anyone used Siemens SIP/Dect phones?
On Sat, Mar 22, 2008 at 09:39:47PM -, Chris Bagnall wrote: Can’t comment on the C460, but the S450 definitely doesn't have these issues: - No SIP call transfer feature (that I can find) Hit ext call during a call, create a new call, then you can SIP transfer between them. Where is that? I don't seem to get that option. What I want is an announced call transfer to another SIP device. - Base station supports multiple phones, but you can only register each handset with one base station. So if you have multiple base stations, you can't take advantage of that feature (i.e, allow the same handset to be used in multiple locations. Other DECT handsets that support multiple bases are available though.) S450 handsets will register to 4 bases. I've got S460IP, that that only seems to allow one base station. - I find it a bit quiet on the sound quality, sometimes a problem with background noise. Should be an option in the web interface to adjust volume - I set all the ones I deploy to high Yeah, tried that. Still a bit quiet when compared to other handsets. -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ -- 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] Which command line is used to send emails to notify incoming voicemail ?
On Fri, Mar 21, 2008 at 12:21:09PM +0100, Olivier wrote:
In Asterisk full log, I can see
Mar 20 14:36:41 DEBUG[29025] app_voicemail.c: Sent mail to
[EMAIL PROTECTED] command '/usr/sbin/sendmail -t'
But when I type /usr/sbin/sendmail [EMAIL PROTECTED] I can't see the same
log lines with this id field.
According to the exim docs (if I understand correctly) the message ID used
is derived from the incoming header:
Message-Id:
So, I assume that whatever is submitting the messages to exim is also adding
this Message-Id: header line.
If there is no message ID, then exim uses its internal message ID, but
doesn't appear to log an id= line.
You could write a wrapper script for your incoming faxes that uses some sort
of date/time+username combination using existing variables available in
asterisk (maybe ${EPOCH} + ${UNIQUEID} + the recipient would do) or
something from the shell script to call the file name.)
R.
--
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510
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Re: [asterisk-users] Anyone used Siemens SIP/Dect phones?
On Sat, Mar 22, 2008 at 09:08:43AM +, Alan Lord wrote: Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the soon-to-become-available-in-the-uk S685IP. Have been using the C460IP phones and they seem to work okay, the range on them is excellent. I haven't had any problems with the base de-registering from asterisk though. (maybe a NAT timeout issue?) They are very simple to configure. Limitations: - No SIP call transfer feature (that I can find) - Doesn't have any remote provisioning features (yet) - Doesn't have any ability to forward calls from the analog side - SIP side, which is a shame as that would be handy. - Base station supports multiple phones, but you can only register each handset with one base station. So if you have multiple base stations, you can't take advantage of that feature (i.e, allow the same handset to be used in multiple locations. Other DECT handsets that support multiple bases are available though.) - I find it a bit quiet on the sound quality, sometimes a problem with background noise. - No VoIP message waiting indicator Maybe some or all of these are addressed in the C475IP? -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ -- 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] Is Asterisk ready for Prime-Time?
On Wed, Mar 19, 2008 at 11:43:21AM -0500, Bill Andersen wrote: This is not a troll. I've used my real email because I want this taken seriously. I'm not trying to make anyone mad, I just want some real discussion on this issue. Please bare with me... 2) Are there any users out there that really DO have an Asterisk system that just works like clockwork? I'm saying, once setup, run for a year (or more) without any issues? 3) If SO, Should I simply consider a different vendor? It depends. As they say, Your Mileage May Vary You have gone with a pre-built asterisk based solution rather than rolling your own with 'plain' asterisk system. So without knowing your particular environment, it's obviously difficult to comment. By the sound of it, your experience of asterisk has been based on one particular integrator's build of it. One or two versions of asterisk out there were lemons and were best avoided. And then there are some modules which are less stable than others. I have found that most of the core asterisk stuff to be reasonably stable and well behaved, but there are a few modules that either have problems, or have had problems in the past, which have now been fixed. chan_agent was a good example of something that worked on a small scale but certain bits of it were just broken. Other problems may be down to operating system, memory, hardware or driver issues. Here, I am using exclusively SIP devices, SIP media gateways (rather than PC hardware) with asterisk voicemail module and seems pretty stable. (We had to reboot the box 9 weeks ago for a kernel security update.) pink*CLI show uptime System uptime: 9 weeks, 4 days, 23 hours, 44 minutes, 22 seconds We have about 77 SIP devices and these are a mixture of hard and soft phones, with four media gateways. Spread over 9 sites. There are a few ongoing intermittent issues, but haven't had any spontaneous crashes so far. 4) If NOT, and if my expectations are that a system SHOULD just run and run without any problems. Is Asterisk simply not my solution. Is Asterisk not REALLY ready for production. Because in my mind (as a user of phone services), dealing with the phone system, even on a MONTHLY basis, means that the system We did evaluate a number of other systems before we decided to go down the route of just plain asterisk and rolling our own, as nothing quite did what we wanted. You could look at OpenSER but I'm not convinced you'd find that an easy thing to work with, when you describe what you want to achieve. SipX was also pretty good, but these are SIP only servers rather than asterisk's multi-protocol ability (You also have to provide SIP media gateways rather than talk directly to a card in the back of the machine) http://www.sipfoundry.org/sipX SIPx is the open source release of Pingtel's SIPEchange product, which I also evaluated. it seemed like a pretty good 'set and forget' solution, and they are also now selling an integrated SIPx appliance: http://www.patton.com/products/pe_products.asp?category=348tab=fb; Which we looked at and was pretty good. Up to 30 users and included automatic handset provisioning, nice GUI for setting things up etc. This is great where you have an environment where running a server is not possible. (our asterisk server is hosted in a nice data air conditioned centre with redundant disks, power, UPS, network.. everything, but no everybody can run an environment for ultra reliable servers, so an Asterisk Appliance might be a way forward and requires no server housing capability and very little knowledge of the operating system etc. It is very difficult to stop thinking 'old PBX', and start thinking What is it we're trying to achieve? If what you want is a PBX, go and buy one. It was a tricky journey from the old PBX system to asterisk VoIP, as there were certain expectations of the old system, and maintaining lots of functionality with the new handsets/asterisk. The system that replaced our PBX doesn't have anything like as many call features as the old PBX did, but then again, most of these features were almost never used. But what we did gain was much more flexibility, choice of handsets/clients, connection to various VoIP networks, the possibility of remote workers, redundancy in the new system, and integration possibilities with existing systems that were completely impossible on the old PBX system. (Or were only possible for lots of money!) Handsets are finally evolving now, trying to put in features that were present on old PBXs with 'traditional' paradigms like key and lamps etc, which users want on VoIP systems, but I believe that will ultimately lead to more proprietary systems and will ultimately fail in favour of Soft Phones, which are much better able to add new features rather than be constrained by a physical handset with buttons and memory limitations etc. In my experience, you can buy a very expensive
Re: [asterisk-users] Newbie Queue: Simple Queue Problem
On Tue, Mar 18, 2008 at 06:20:02PM +1100, Lee, John (Sydney) wrote: I am trying to build a simple queue for the receptionist phone. In other words, there is only 1 agent and that is the receptionist phone. However, when I call from an outside line to another extension which I then forward to 4000, I cannot get into the queue. exten = 98786983,1,Answer() exten = 98786983,n,Dial(SIP/4000,20) exten = 98786983,n,HangUp() SIP devices defined in sip.conf do not magically become extensions in extensions.conf by virtue of them being there. i.e, a dialplan (extensions.conf) entry of 4000 bears no relation to the SIP device [4000]. You just happen to have called them the same thing. Therefore, your: exten = 98786983,n,Dial(SIP/4000,20) Is routing to the SIP device 4000 rather than the queue 'console'. So you either need to go a Goto(context,4000,1) or to drop it to the queue with Queue(console) etc. R. -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ -- 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] Interrupt VM and Steal a call.
On Fri, Feb 22, 2008 at 09:05:17AM -0500, Michael Munger wrote: Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're leaving it? I imagine that would be tricky to do once the call has been handed in to the voicemail application, as presently you are limited in what you can do to the call once it's gone in there. You might be able to locate the original SIP channel and bridge the calls, but I've no idea how you would track that properly. There is a way to make voicemail have a press 0 to be transferred to somewhere else option. We use that here and it works. Users can set up where they want the caller to be transferred to (usually a mobile) and then they can record on their outgoing message leave me a message, or press 0 to try my mobile... Or, Sounds like a case for a few IP DECT cordless handsets to save all this running about! You might run into somebody carrying a boiling hot cup of coffee in your rush to answer the phone! (happens!) We have a few Siemens C460 IP DECT Phones. The range and battery life on them is by far superior to any of the WiFi/SIP phones I've tried so far. I have a SIP/Wifi Nokia E65 that works great, but the battery life is not very good when the wifi is left on, and it was less than straightforward to set up! 2. If a phone rings a receptionist desk, and the receptionist is down the hall, she wants to be able to dial an extension, and have that transfer the call from her desk to the phone she's currently on so she doesn't have to run to her desk. Is there a built in feature for this or do I have to code it? There is a feature called pickup defined in features.conf: pickupexten = *8 Restart asterisk if you need to change features.conf (in my experience just a reload when changing features.conf doesn't always work) You then need to define your SIP/devices into pickup groups in sip.conf, for example:- [500] canreinvite=yes nat=no secret=... dtmfmode=rfc2833 callgroup=1 pickupgroup=1 [501] canreinvite=yes nat=no secret=... dtmfmode=rfc2833 callgroup=1 pickupgroup=1 Then reload. Now, if extn 500 were ringing, picking up 501 and doing *8 will connect that call. For busier systems I believe there is a dialplan feature that enables directed pickup so you can pick up a specific extension, but I haven't played with that so I can't say how it works. That might be more suitable. The callgroup defines what pickup group the device is in, and pickupgroup defines what groups (when that extension dials *8) that device can pick up. A device can be in one callgroup but multiple pickup groups:- pickupgroup=1,2 This is so that if you have many sites or departments, only people who sit within the range of the ringing phone can pick it up, and not get connected to some other random call incoming somewhere else. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ -- 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 - Is handover included in DECT GAP ?
On Thu, Jan 10, 2008 at 11:22:29AM +0100, Olivier wrote: Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Yes. It is a capability of the handset, where, as the user is moving about, the handset is continually scanning for the best channel/frequencies available from the base station. If a better signal than the one the user is currently using becomes available, (perhaps they have gone into a different room or moved behind a tree etc. meaning the current channel is weaker, then the handset will switch to using the better channel during the call.) There are a few systems out there that support multiple base stations which, to the handset, all look like the same registered base. (i.e, the handset only registers/authenticates once with the system (and not every base station) and then as the user moves about the handset, by virtue of always looking for the best channel, will hop from one base to the next. It does require that the system in the middle manage the database of registrations etc.) I've seen such capability on Siemens HICOM and Bosch PBXs, for example. All the DECT base stations are wired back to a central card in the system. I don't know if there is a standalone DECT IP offering supporting similar. The timing/clocking to support seamless roaming between the base stations is complicated and has to be very precise, so I imagine that such a system would need to have one central controller (and the SIP gateway function) with DECT base stations all wired out from there, rather than lots of independent DECT bases with Ethernet, that talk to a central unit over IP and somehow hand off the call. So to span multiple buildings you would probably need dedicated copper pairs or fibre to connect in the remote base stations to the central system. That is certainly the way it worked when I was last tinkering with DECT stuff. Although the Siemens switches could have multiple remote shelves connected over fibre to different buildings, the DECT bases all had to be connected via copper cables (and be powered on by) a central card in the main shelf, and could not be connected to a card in a remote shelf. (Or, you could have multiple PBXs and handsets roaming between different systems, but that starts to get expensive for maybe 10 users!) I have some detailed specs on it somewhere if you want more technical info. Rob ___ -- 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 - Is handover included in DECT GAP ?
On Thu, Jan 10, 2008 at 01:47:39PM +0100, Michiel van Baak wrote: We have 2 different setups in production. NEC-Philips ip-dect and the kirk/tiptel ip-dect. The NEC-Philips one works with a dedicated server to controll the registration etc, and all the radios are connected to the normal ethernet network. No need for dedicated copper/fiber, they simply communicate over the lan with the central provisioning/managing server. The handsets register with the closest radio and from that moment on they can roam to all radios. Asterisk sees every handset as it's own sip entry. Interesting. Does that support handover between bases during a call, or only registration to nearest base station? Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ -- 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 - Is handover included in DECT GAP ?
On Thu, Jan 10, 2008 at 02:40:21PM +0100, Olivier wrote: that is what I call handover roaming = without any ongoing call handover = with ongoing call this would need the appropriate logic in the base stations. I know such hardware exists (Kirk!?), Kirk base stations support roaming and handover but it's very difficult to know which handsets, beside Kirk handsets, support such feature as vendor won't specify if Kirk base stations don't support Siemens handsets, for instance, because working feature set is very poor or because we don't want to care or deal with non-Kirk handsets as we sell our own. As far as I understand it, where you have a base station and one phone, you are creating a cell consisting of one antenna. It is possible with DECT to have one cell consisting of many antennas (base stations) Moving between base stations in the same DECT cell is called handover. The handset registers with any base in the same cell. If you only have one base station, then the base station and cell are effectively the same thing. The base stations have to be connected somehow to a central system that manages the handover in the cell during a call, the system has to support multiple base stations in the DECT cell. No additional functionality should be needed in the handset, because it already has the capability to channel hop between DECT channels during a call, without the user noticing. Many DECT handsets (but not all) support registration with multiple DECT cells (i.e. as well has having up to X handsets registered with the cell, a handset can register with up to X bases.) This would support 'roaming' in the sense that the handset would use the base it could register with. (i.e, if you had one DECT handset registered with a base at home and at work, the handset would work on both bases if the handset supports multiple bases.) This could be called roaming between DECT cells and is done by the handset, but handover between two completely different DECT cells during a call is not possible. In a large PBX installation, you can have both working, so that if a user took a handset from one system in one city, to another office in another city, and these PBXs were connected together, the Home PBX receives registration request from the remote PBX, and diverts the calls for that user to the remote PBX. I believe this is done by proprietary vendor software magic though, and is not part of DECT itself. Rob ___ -- 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] Sounds transscript / speech synthesis
On Sat, Dec 22, 2007 at 12:55:19PM +0100, Jay R. Worthington wrote: Hi, in the earlier version there was a sounds.txt with the transcript of the soundfiles. Does this still exist somewhere? Not an up to date or complete one, the last time I looked, so I ended up transcribing them all so that we could re-record them into UK English. We adapted a few of the phrases so as not to confuse UK users. Instead of pound we say hash and instead of Password we say PIN as some users had no idea what was meant by hash or password, and since they have SIP passwords and login passwords, this was just another password we could do without confusing them with! I can send you a copy of the document if you want. There are over 300 individual sounds with a few extra that we added. Even then I've had complaints from users that the voicemail menus are too long-winded and although functional, it isn't the easiest of systems to use compared to others. (We re-recorded the vm prompts pretty much word for word, but shortened a few of them, so for example: To exit voicemail press the pound key. ended up as: To exit, press hash. The next time we do any recordings, we may record it in a better style, as the default sounds are considered bad practice by some(?): Press 1 to record your unavailable message. Press 2 to record your busy message. Press 3 to record your name. Press 4 to record your temporary greeting. Press 5 to change your PIN. Press star to return to the main menu. I want to re-record it so that press x is after the option, so when you hear the option you want, if you were not paying attention, you don't miss the number you were supposed to press, and you don't have to listen to all the options all over again, so: There are five options. To record your unavailable message, press 1. To record your busy message, press 2. To record your name, press 3. to record your temporary greeting, press 4. To change your PIN, press 5. To return to the main menu, press star. Experience shows that users seem to prefer it this way around. There are some inconsistencies between the various sound sets, for example, some menus, press star for help implies that there is further help. What they actually mean is to hear the menu again, press star In other menus, * goes back up to the previous menu, or does nothing. In some menus we get told to dial and others press. Some options require pressing the # key after, and others not. We also edited all the embarrassing I am soorry... type announcements, or removed I whenever it appears. Machines cannot relate to callers as I. It's all a bit fake and insincere. The machine cannot be sorry Callers here find these sort of I'm not really sorry, I'm a machine announcements annoying and somewhat patronizing. (Using a recorded announcement to try and get 'personal' and 'friendly' just doesn't work; it's clearly a recording and couldn't care about your feelings one way or another, so don't insult my intelligence by pretending it does.) Here in lovely England, an American woman saying I am soorry.. provokes anger and the response along the lines of *STOP TELLING ME YOU'RE SORRY AND JUST F*((*! DO IT! followed by slamming the phone down/pounding of fists on the desk! I kid you not. It's probably just a cultural thing, but I wanted callers not to get annoyed by the system, so we embarked on the mammoth 5 hour task of re-recording every single sound, as the small group of pilot users we had before the main system went live, all hated the default sounds! Is there a plan to make speech synthesis available the same way as soundfiles, ie. instead of playing language/soundfile.wav, send the text to the speechengine and play the output...? I've heard one or two systems using this, but it sounds a bit strange in my opinion, and sometimes rather difficult to hear. I don't know if there are any plans to change the way asterisk plays sounds (I imagine getting it to say everything correctly in many languages would be a long and complex challenge!) I'd sooner spend some time recording the sounds rather than spend ages listening to the output of the text to speech system and tweaking phonetics to get the intonations/accent right. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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 7960 soft key customization?
On Mon, Dec 10, 2007 at 10:06:02AM -0500, Peter Pauly wrote: Does anyone know how to customize the order of the soft keys on a 7960 running SIP? All the documentation I could find is CallManager related. Specifically, I want to move the transfer function to the first set of buttons during a call. I looked in to doing just that, as users complain that transfer is fiddly. Conference should be on the second screen since it is used less, but Transfer is on the second screen, but used all the time. As far as I am aware, there is no way that these soft keys can be moved. Oddly, Cisco have addressed this in the 7961, as they appear to have removed the BlndXfr feature (just use Transfer and then hang up to do the same thing) but the 7961 runs completely different software to the 7960. The short answer is: No, it can't be done. Why? The slightly longer answer: Note that the Cisco SIP only image does not support XML push, or many of the more advanced features. (SoftKeyItem, for example does not work.) as these phones only support an earlier version of the XML SDK (v2) and not v5 like the 7961, which supports things like start URL. I'm not even sure you could move a builtin feature with that anyway. The 7960 / 7961 phones running SIP are good. The interface is nice and clear and easy use and easy to read, unlike some other handsets. It has great potential but is spoilt by some annoying foibles like this. Sadly there appears to be no incentive for Cisco to develop the SIP image as apart from handset sales, they don't make any money out of it. As far as I understand it, the SIP image is provided as a piece of fluff purely to allow customers to roll out Cisco phones on a SIP platform, and then the dual-boot Universal Application Loader allows migration away from a competitor's (SIP Based) system, as you just boot the phone up in CCM and it puts on the right version of software. If you were a large corporate rolling out 3000 Cisco 7960s but only had one Ethernet port per desk, maybe you can't have two IP handsets on every desk, and integrating the two phone systems back-end would be a time consuming nightmare/expensive just purely to rip it out again a few months later, and you probably couldn't swap out 3000 handsets overnight, or even in one weekend. Enter the SIP image to assist in migrating away from the SIP interface of your old system, to the lovely shiny Cisco CallManager you just bought. Of course, you could in theory use the SIP image/dual boot to migrate away from CCM to a competitor's system, however, what would happen is that users would lose a whole bunch of functionality they had with CCM that doesn't exist in the SIP image, and so the new system would be 'worse than the old one' from the user's perspective. When you see it like that, you can see that Cisco couldn't really care less about the little guys running asterisk on their migratory SIP image, which they only created so they can provide a migration path to big corporates away from their competitor's systems. Every so often I think that there must be a better handset out there, and indeed there are better handsets out there that allow things like call reject, Busy lamp field etc. (SIP feature-wise the Cisco phones are very basic.) but the interface tends to be quite complicated/weird compared to the 7960. I was very impressed with the aastra, but if I found the interface a bit too complicated compared to Cisco, the users would have a nightmare! Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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] Odd bug in Siemens C460IP ?
On Sun, Nov 25, 2007 at 01:10:08PM +0100, Olivier wrote: Could you get from Siemens some kind of commitment to fully support Alert-Info or at least, to ignore Alert-Info data in incoming INVITEs ? Siemens responded to my initial query yesterday with a rather unhelpful: 'It is an asterisk issue, and we don't support asterisk' type response. But they did point me to a link: http://www.voip-info.org/wiki/view/Siemens+Gigaset+C450IP On this page, somebody suggests a workaround: EDIT: Actually, there is a way to have alert_info work with other phones and still have the C450IP work with the same configuration. Even though the C450IP will not have a distinctive ring. Just make sure that you set your alert_info variable in the following way: exten = s,1,SetVar(_ALERT_INFO=something). Note the and in the command. The C450IP will still ring, and at least the other phones I have tried with will accept that syntax. I have not tried this so I don't know if it works yet, as not sure yet if it will break other things or which is valid syntax for the SIP header. I have a sneaky feeling that this method of SetVar(_ALERT_INFO is replaced with SIPAddHeader() only in asterisk 1.4 anyway? My experience with Siemens in the past is that they will probably not fix a bug like this unless you are negotiating a large contract and it becomes a critical item for them to address in order for them to get the business. Individual joe-users usually don't count for much. Otherwise they just seem to stall and stall, and then eventually phase out the particular model of handset and replace it with a completely different model, usually with many of the same bugs and foibles. They *do* seem to have addressed one pet hate I had of all the Siemens DECT phones though, in that finally the Red and Green call answer/hangup buttons are positioned with the green button on the left, and the red button on the right, and not the other way round, like pretty much every other mobile/cordless handset on the planet. (the times I went from Nokia handset to Siemens handset and out of habit rejected calls instead of answering them!!) Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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] Odd bug in Siemens C460IP ?
Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten = s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a different ring cadence so to differentiate between external and internal calls. Other handsets that do not support Alert-Info: just ignore the presence of this header. When this header is set in a call to the C460 IP, it does not alert, in fact it does not respond to any INVITE requests; asterisk just retries the requests a few times and then gives up. Anyone able to reproduce? I have firmware version 0107 / 041.00 I suppose as a workaround I could add an astDB entry for these extensions, and a bit of logic in the dialplan to tell asterisk not to add the header for extensions that have that flag set. Regards, Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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] Changing audio message to text message
On Fri, Nov 16, 2007 at 02:28:45PM +0100, Anthony Chapellier wrote: Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? You might need to clarify what you are trying to do. When a call comes in for a particular queue, instead of playing an audio message in-band at the caller please hold the line you want to send some sort of text message somewhere... What sort of technology do you have in mind that you want to integrate? SMS? URL messages to other IP handsets? CTI integration with a web browser? pop-ups on user screens? Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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] Pass CallerID when call forwards to PSTN?
On Thu, Nov 15, 2007 at 06:18:33PM +, Russell Horn wrote: Hi, Incoming calls to one of my lines are set to ring two internal lines and simultaneously start ringing my cell phone. Something like this: exten = s,1,Dial(SIP/2201SIP/2202IAX2/[EMAIL PROTECTED]),90) The internal lines 2201 and 2202 will both see the callerID for the incoming call, but my cell phone will show the callerID for asterisk, not the calling party. What's the best solution to take the callerID from the inbound call and transfer it to the outbound one? I think your carrier has to permit you to set callerID to something that is not one of your numbers in the range you have been allocated. Some carriers allow it, and others do not. (To prevent end-users from being able to forge the caller ID to anything they want!) I've never figured out if it's possible to get BT to do this for our ISDN lines, I think I tried to get them to allow it, but I think they said we have to have an OFCOM licence or something... ___ --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] Pickup Command not working
On Tue, Nov 06, 2007 at 05:04:50PM -0500, Lutgring, Sam wrote:
When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE
603. I am dialing **212 with the following config. Anyone have a
suggestion?
I am not sure, but in the context where your extensions are, have you done:
include = BLF_Group_Pickup
EXTENSIONS.CONF
-snip-
[BLF_Group_Pickup]
; Defines how the extension to pick up a ringing phone in your BLF group
exten = _**XXX,1,Pickup(${EXTEN:2})
exten = _**XXX,n,Hangup()
[BLF]
; Defines a BLF Hint for phones
exten = 212,hint,SIP/sam
-snip-
This bit should have some context= i,e, where your clients are dialling, do
they have access to the [BLF_Group_Pickup] bits of the dial plan?
(I think it can also be set as a default in the [general] part of sip.conf
i,e, context=default if not defined in the SIP peer config, but for
security reasons, your internal clients should ideally be in a separate
context, so you can differentiate between internal and external connections
and limit what they can dial.) Mine is called from-client and then in
each [client] section in sip.conf, I have context=from-client
You might also have to set the Pickup() command to pickup from the correct
context, i.e. Pickup(${EXTEN:[EMAIL PROTECTED]) if it still doesn't work.
SIP.CONF
-snip-
[sam]
type=friend
username=sam
fromuser=sam
callerid=sam
host=dynamic
dtmfmode=RFC2833
disallow=all
allow=ulaw
call-limit=20
subscribecontext=BLF
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Re: [asterisk-users] 7960 Queue Issue
On Mon, Nov 05, 2007 at 12:09:48PM +1100, Nick Brown wrote: Thanks Eric, this is the case. A bit of a shame that it removes the functionality for the member to see calls that have not come from a queue however there is not much choice in the matter. It works for me... somehow... I have Cisco 7960 phones also. I think I add the Local/xx instances into the queue instead of the SIP/ device names, and then have a context that checks the state of the SIP channel before trying to place a call to it. (So, member = Local/[EMAIL PROTECTED]) where agent_call is the context to go to in the dialplan that handles the agent calls (and passes it to another queue/voicemail if the queue drops out with full/unavailable etc.) [agent_call] does some stuff with ChanIsAvail checking if the channel is free before placing a call, and if it is found to be busy, it returns goes to a step which returns Busy() which causes the queue processor to move on to the next person in the queue. (It will go to agent_call again for the next destination, and so on.) That way, users can have DDI numbers with call waiting functionality enabled on the handset if they wish, but for queue calls, it goes to the next available queue member rather than stacking up all the calls on one phone. What I have is a simplified (and 1.4/1.2 compatible) version of Example 2 at: http://www.voip-info.org/wiki/view/Agents+without+agent+channel (just look in the [agent_call] bit of this, and you'll see it is using ChanIsAvail to check the status.) I did not need all the functionality of this example, so removed a bit of it, but used it because encountered a few limitations with chan_agent which meant I couldn't use Agents, so replaced the functionality in dialplan logic. (which was bit difficult to do, but it works!) I can send you what I have if you like, but my dialplan is quite complicated as the setup here allows 'agents' to log in and out from any phone, so the users extn numbers are essentially portable. (i.e, the handsets have some meaningless (to the user) extension like 42105 and the user logs in as 710 from that handset. Some database work is done when they log in to map 710 - SIP/42105, fix the outgoing caller ID, and add them to their queues. Alternatively, you might be able to use Agents, but I really cannot recommend it, as for me, it caused more problems than it solved (problems with call waiting, transfers, and the fact that the feature it relies on, AgentCallbackLogin() is deprecated in 1.4 anyway. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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 delete voice mail messages?
On Mon, Nov 05, 2007 at 12:47:52PM +0100, Michiel van Baak wrote: On 12:15, Mon 05 Nov 07, voip crazy wrote: Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any messages index file or there isn't any file to index them? Could you share any script to do that? Hi, Voicemails are stored in /var/spool/asterisk/voicemail/context/vmbox by default. There's some .wav files and a .txt file for every message. You can easily delete them using some shellscript. Yes, but you must not just barge in and start deleting them, they have to be renumbered in sequence after you delete the ones you want, otherwise the vm app breaks when the user is listening to their messages. I think there is also a way to lock the files (I think with .LCK files) so that the vm app does not try to write them while you are manupulating. (and so your script can detect that there is a message being created.) I expect you will be able to find some code out there that does it without breaking it. (vmspool_manager) ? Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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] Ringing Groups, SIP Forward and looping problem
On Fri, Sep 28, 2007 at 09:57:52AM +0100, Russell Brown wrote:
I've a big problem with SIP forwarding back into 'ringing groups'
creating what can only be described as call storms :-(
I have a 'ringing groups' of SIP phones with an effective dialplan (much
simplified) like so:
; Purchase ledger
[ptsn_inbound]
exten = _846061,1,Dial(Local/[EMAIL PROTECTED])
I am not sure why you are doing it like this but it seems awkward.
Relying on handset diverts seems fraught with danger as you can't be sure
what's going to happen from a dialplan perspective.
Why don't you set up a queue in queues.conf strategy ringall:
[purchase]
; Dynamic group for users logging on in London Office
strategy = ringall
maxlen = 1
retry = 1
timeout = 20
musiconhold = default
joinempty = strict
leavewhenempty = yes
timeoutrestart = yes
member = SIP/110
member = SIP/111
member = SIP/112
member = SIP/113
member = SIP/114
Then route calls to that queue from the dialplan:-
exten = _846061,1,Queue(purchase|rn|||40)
...
[...variety of options you can do here if there is no answer all busy
in the queue etc, see variable ${QUEUESTATUS}. Here's what I've got:-
exten = s,n,GotoIf($[${QUEUESTATUS} = UNKNOWN]?200)
exten = s,n,GotoIf($[${QUEUESTATUS} = BUSY]?200)
exten = s,n,GotoIf($[${QUEUESTATUS} = FULL]?200)
exten = s,n,GotoIf($[${QUEUESTATUS} = JOINUNAVAIL]?200)
exten = s,n,GotoIf($[${QUEUESTATUS} = LEAVEUNAVAIL]?200)
exten = s,n,GotoIf($[${QUEUESTATUS} = LEAVEEMPTY]?200)
exten = s,n,GotoIf($[${QUEUESTATUS} = TIMEOUT]?200)
]
Then you could set up some features in the dial plan to allow your users
to go in and out of the group as required. Something like:-
exten = _*71,2,Macro(togglegroup,${CALLERID(num)})
( *71 will toggle in and out of group, so you could program a button on
your phones for example, to set them in and out of group. This set of
macros keeps track for each user in and out group state and toggles
it in and out. It keeps track of it with a db variable.)
[macro-outofgroup]
exten = s,1,NoOp(macro-outofgroup reached: ${ARG1})
exten = s,n,NoOp( -- DND pausing queue member: Local/${ARG1} --- )
exten = s,n,PauseQueueMember(|Local/[EMAIL PROTECTED])
exten = s,n,Set(DB(${ARG1}/outofgroup)=1)
exten = s,n,Answer
exten = s,n,Playback(extras/dnd-out-of-group)
exten = s,n,Hangup
[macro-ingroup]
exten = s,1,NoOp(macro-ingroup reached: ${ARG1})
exten = s,n,NoOp( -- DND unpausing queue member: Local/${ARG1} --- )
exten = s,n,UnPauseQueueMember(|Local/[EMAIL PROTECTED])
exten = s,n,DBdel(${ARG1}/outofgroup)
exten = s,n,Answer
exten = s,n,Playback(extras/dnd-now-in-group)
exten = s,n,Hangup
[macro-togglegroup]
exten = s,1,NoOp(macro-togglegroup reached: ${ARG1})
exten = s,n,GotoIf($[${DB(${ARG1}/outofgroup)} = ]?900)
exten = s,n,Macro(ingroup,${ARG1})
exten = s,n,Hangup
exten = s,900,Macro(outofgroup,${ARG1});
exten = s,n,Hangup
(I've got those sounds if you want them, let me know, if you don't mind
plummy british accent we re-recorded all our sounds files in, plus a few
custom ones, or you could just play a tone so the user knows the group
action has been carried out.)
Let me know if this is any use to you.
Regards,
Rob
--
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sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510
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Re: [asterisk-users] Ringing Groups, SIP Forward and looping problem
Whoops! Forgot to change it for SIP devices.
Of course you need to change your queue member devices to SIP and not
Local/${ARG1} as I've got agents and other complications in mine.
You might need a context or not, see what happens!
Rob
Here is corrected version (I think will work, untested though!)
[macro-outofgroup]
exten = s,1,NoOp(macro-outofgroup reached: ${ARG1})
exten = s,n,NoOp( -- DND pausing queue member: SIP/${ARG1} --- )
exten = s,n,PauseQueueMember(|SIP/${ARG1})
exten = s,n,Set(DB(${ARG1}/outofgroup)=1)
exten = s,n,Answer
exten = s,n,Playback(extras/dnd-out-of-group)
exten = s,n,Hangup
[macro-ingroup]
exten = s,1,NoOp(macro-ingroup reached: ${ARG1})
exten = s,n,NoOp( -- DND unpausing queue member: SIP/${ARG1} --- )
exten = s,n,UnPauseQueueMember(|SIP/${ARG1})
exten = s,n,DBdel(${ARG1}/outofgroup)
exten = s,n,Answer
exten = s,n,Playback(extras/dnd-now-in-group)
exten = s,n,Hangup
[macro-togglegroup]
exten = s,1,NoOp(macro-togglegroup reached: ${ARG1})
exten = s,n,GotoIf($[${DB(${ARG1}/outofgroup)} = ]?900)
exten = s,n,Macro(ingroup,${ARG1})
exten = s,n,Hangup
exten = s,900,Macro(outofgroup,${ARG1});
exten = s,n,Hangup
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Re: [asterisk-users] Limiting Simultaneous calls
On Wed, Sep 19, 2007 at 01:56:42AM +0530, Jim Boykin wrote:
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
I think you can do this sort of thing with the Set(GROUP) and GROUPCOUNT to
monitor number of calls placed in a call 'group' which in this context does
not mean a pickup group or a caller group, it means 'a group of calls set up
in group $foo' (where $foo is some variable)
Take a look at:-
http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
and:-
http://www.voip-info.org/wiki/index.php?page=Superdial%20macro
To see how it is used to limit the number of outgoing calls to a PSTN
carrier.
'group' could be a global setting you give it, or the extension number of
the user (to limit globally or per extension)
Specifically:-
${ARG6} - Max. group number (maximum number of concurrent calls you want to
allow for that group)
exten = s,1,Set(GROUP()=${ARG5})
exten = s,2,Set(GROUPCOUNT=${GROUP_COUNT(${ARG5})})
exten = s,3,GotoIf($[${GROUPCOUNT} ${ARG6}]?104)
exten = s,104,Goto(s-CHANUNAVAIL,1)
etc.
Rob
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Re: [asterisk-users] Queue agents w/ DUNDi
On Tue, Sep 18, 2007 at 11:27:36AM -0500, Kyle Sexton wrote: All, I'm trying to configure queue agents w/ a DUNDi setup so that an agent can login to whatever server they please w/o any custom setup. In general this seems to work, agents login w/ AgentCallbackLogin into the incoming context (not a special queue context) and can receive queue calls. Don't use AgentCallbackLogin() it's odd in some interesting ways (The whole agent stuff isn't very flexible in many ways if your users have multiple ways to get called outside of the Agent.) For example if you have users in queues represented as Agents with also direct numbers respresented as SIP/xxx elsewhere, you will have problems with call waiting and busy detection not working properly, i.e, when the user is making an outgoing call on their SIP extn, the agent stuff does not detect them as being busy, so you cannot use call waiting. An 'agent' can only accept one call at a time but SIP/xxx may have several calls. About your situation, you might be able to solve it by using Local/[EMAIL PROTECTED] to route the call to where you need it to go when a call comes in for an agent that you want to locate in the dialplan somewhere else. The thing you route to using Dial(Local/xxx must be something in the dialplan routable by the current context.) AgentCallbackLogin as I understand it, deprecated as of 1.4.x, and 1.2.x is no longer being actively developed, so I'm trying to get off it, however some stuff I do is not possible now without that feature that they don't seem all that concerned about fixing right now. :-( Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ 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] Cisco 7960 or 7960G
On Sun, Sep 02, 2007 at 03:47:45PM +0100, Chris Bagnall wrote: There's both a 7960 and a 7960G (and a 7961 to confuse matters further). The 7960 is the earlier version. The easiest way to identify it from a picture is to look at the messages/services/etc. buttons. On the 7960 the words messages and services are written on them. On the G, there's an envelope and a globe on the buttons themselves, and the words messages and services are provided on a surround sticker (one assumes to make internationalization easier). ...although I don't think Cisco ever produced any other languages for the 7960G anyway, but 7960 and 7960G are pretty much identical. 7961 is a completely different phone with totally different software, although it has a better screen and much better audio quality than the 7960. 7960 was end-of-life a while ago by Cisco. Not sure about the 7960G though. If you run them in SIP Only mode, they are quite limited when it comes to actual functionality when compared to what other phones are offering. 7961, although a better bit of hardware, does not offer much noticable improvement for SIP. The functionality is about exactly the same, but with more possibilities for integration via XML than the 7960. 7961 does support standard 802.3af PoE and not Cisco's legacy proprietary PoE system which they introduced before 802.3af. You need a Cisco switch or a switch that supports legacy PoE (Foundry FES for example) to make the 7960s power on, but 7961 works with standard 802.3af PoE kit. Contact me off-list if you want my list of specific limitations of the 7960/SIP, as there are many. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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 can i send my sip channel 3 to mailbox 2? Please Help!
of extensions.conf required. Just some food for thought on what is possible. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --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] off-hook warning tone
On Sun, Sep 02, 2007 at 05:15:41PM -0500, Anthony Messina wrote: is asterisk capable of generating the off-hook warning tone for the us? 1400+2060+2450+2600/100,0/100 i have placed it into indications.conf, but all i get is one high-pitched screech instead of alternating tones. I am thinking this might be handset specific thing, as unless you dial something the call is not going to be placed to asterisk yet, unless you can somehow first Answer() the call after some timeout (i.e, if the handset has a hotline extn config to dial after N seconds of no digits being dialed - some handsets support that functionality) Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 pgpR7r3AeW3j9.pgp Description: PGP signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] off-hook warning tone
On Tue, Sep 04, 2007 at 07:41:53AM -0500, Anthony Messina wrote: well i'm looking for the feature that the telco provides where, if you've left the phone off-hook for 60 seconds or so without input, it gives you the loud put the damn phone back on the hook noise. it works if i set absolute timeout to 60 and use the congestion tone, but i was hoping to use the actual off-hook warning tone. it seems as if the tone itself is not generated properly within asterisk. Curious as I have not had problems with generating the tones. It's worth checking that in sip.conf the language= option is set to the same section you are editing in indications.conf In the dialplan, what I think should happen is that when you do: Congestion() You send a congestion message back to the phone using SIP (rather than in-band audio) so the handset is probably generating the Congestion tone, not asterisk as it is not yet in the media path. If you did it inband audio:- Answer() Playtones(congestion) This would play the tone from indications.conf - have an experiment with this by setting up a little extension and dialling it. As far as I can tell, AbsoluteTimeout() is just a global timeout for the duration of a call, so if you set it to AbsoluteTimeout(30) then the call (any call) will be hung up after 30 seconds. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AbsoluteTimeout Some handsets allow you to customise the tones played - depends on the handset. And some handsets have a hotline feature to dial a given extension after no digits have been dial for N seconds. (So you could get the handset to dial a special extension which then answers the channel and plays the noise you want!) I could be wrong of course. Never wanted to do this as our phones just seem to go back on-hook regardless after some dial timeout has elapsed. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 pgplaQjJlUl1F.pgp Description: PGP signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to push callerid for each user from sip phone on one side through asterisk (Digium) to E1 card running application on other side
On Thu, Aug 09, 2007 at 11:07:50AM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am running asterisk PBX ( digium TE120P card configured) on one
system. It is connected to E1 card running application on the other system.
After establishing sync between two card, I am able to place call from sip
phone to E1 card running application. I want to pass the callerid, when
calling from sip phone to E1 card running application. Which all
configuration files is to be changed in the asterisk.
I am doing the following changes in extensions.conf
exten=115,1,SET(set(CALLERID(num)=2)
exten=115,2,Dial(ZAP/g1/115,20)
So, when dialling from sip phone to extension 115
it pushes the callerid hardcoded for that extension to E1 card running
application, not for each user in sip.conf.
Can anybody tell me how to insert the callerid to each users? Which all
are the configuration files, where changes are to be made? So that, when I
call from sip phone through asterisk PBX to E1 card running application,
callerid for each user from sip phone called should be forwarded to E1
card running application side. thanks and regards sanchal
There are two ways, depending on your setup.
Easiest method if you have a straightforward SIP config, set, for each
extension in sip.conf, for example:
[1234]
callerid=Fred User 2
...
Then this callerID string should be used.
(This sounds like what you have at the moment)
If you need to change it to something different depending on the trunk being
used etc there are a variety of ways to do it, depending on how many users
etc. You could use an asteris db lookup instead before you place the call in
extensions.conf to overwrite what is set in sip.conf, to translate SIP extn
caller id to something else (in the right place in extensions.conf):
exten = 115,1,Set(CALLERID(number)=${DB(${CALLERID(num)}/callerid)}}
exten = 115,2,Dial(ZAP/g1/115,20)
Then write a db entry for each client from the CLI:
asterisk -r
asterisk*CLI database put 1234 callerid 2
Updated database successfully
asterisk*CLI database show 1234
/1234/callerid: 2
In this example a call with the callerid of 1234 would get changed to
2 for that call.
This would be a good approach if there was no apparent relationship between
1234 and 2.
There are ways to modify the callerID on the way out based on the extension,
say for example you have a callerID of 43703 and you just want to translate
that to 703 on the way out, you could extract the digits and replace.
In this example if the callerid is in the range I want, translate the
callerID to something else (in fact we just take the last three digits)
exten = 115,n,ExecIf($[$[${CALLERID(num)} = 43000] $[${CALLERID(num)}
= 43999]],Set,CALLERID(number)=${CALLERID(num):-3})
Hope one of these answers gives you some inspiration...
Rob
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Re: [asterisk-users] Asterisk 1.2.18 problem
On Sun, May 27, 2007 at 05:43:59PM +0200, MOSBAH ABDELKADER wrote: hello, I have installed asterisk 1.2.18 in suse 10.2. After typing asterisk in the terminal command line (i don't think that asterisk runs when doing this) i type asterisk -r but the response is Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?). Is asterisk running? If it is not running (i.e, configuration file missing somewhere) then you need to correct that. Check the permissions on the file /var/run/asterisk.ctl. If you are running asterisk -r as a non-root user, then you need to make sure that user has permission (group etc.) to read/write this fiel. Rob ___ --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] Cascading Queues
On Thu, May 17, 2007 at 03:50:52PM -0400, Jason Adams wrote:
Scenario 1:
We are working with a client that currently has one support queue with
about 10 agents. They are starting to get pretty long hold times for
their customers and they have requested three queues. Queue 1 will have
a timeout of 4 minutes. After that it will move to Queue 2 which has a
default timeout of 3 minutes. After that we will transfer the call to
the receptionist who will either take a message or put them back in the
queue with a higher priority if they want to continue to hold. Queue 2
will have more agents in that queue plus the agents that were in Queue
1.
Question:
Can I have the same agents in multiple queues to work the way I'm
thinking above? So if the caller reaches Queue 2 the agents from Queue
1 will be available if they get off a call, plus new agents are added
into Queue 2.
You can have the same agents in multiple queues, chan_agent only allows one
call to happen per agent channel. As long as you are not mixing Agent/xxx
and SIP/xxx destinations which are routing to the same people in the queues,
it should work. (When an agent channel is busy, the SIP channel might not
be, and vice-versa.)
So I'm thinking something like:
exten = s,1,Queue(support1)
exten = s,2,Queue(support2)
exten = s,3,Dial(SIP/${RECEPTIONIST})
Then the receptionist would just dial a special extension which would
add priority=10 to the queue.
You might want to check the ${QUEUESTATUS} so you can work out why the call
dropped out of the first queue, and if you want it to immediately drop out
of the first queue if it is full, or sit there for 4 minutes waiting etc.
(In queues.conf, check the joinempty/leavewhenempty options for the queue,
then check it like:- (where the numbers are priorites you want to jump to)
exten = _s,n,GotoIf($[${QUEUESTATUS} = UNKNOWN]?500)
exten = _s,n,GotoIf($[${QUEUESTATUS} = BUSY]?850)
exten = _s,n,GotoIf($[${QUEUESTATUS} = FULL]?850)
exten = _s,n,GotoIf($[${QUEUESTATUS} = JOINUNAVAIL]?650)
exten = _s,n,GotoIf($[${QUEUESTATUS} = LEAVEUNAVAIL]?650)
exten = _s,n,GotoIf($[${QUEUESTATUS} = LEAVEEMPTY]?650)
exten = _s,n,GotoIf($[${QUEUESTATUS} = TIMEOUT]?700)
Your setup will need a bit of work, for example what will happen if the
receptionist is not available, how to trap calls going round and round in
loops etc. You could automate bits of this further. (Maybe with the
voicemail app and a breakout 'o' extn, and record a greeting that says
leave us a message, or press 0 to continue to hold...)
Scenario 2:
This same customer is starting to sell their product internationally.
They are purchasing VOIP DID's from various countries for local calls
from that area. Would this just be like setting up a regular VOIP line
to register the account in sip.conf and then creating a context for
those countries so we know where they are coming from?
Yes. The sip.conf entry for the peer will point it to the right context=,
and/or the register= statement can point to a specific extension, so you can
tell where the call is coming from either way.
If you want to place outbound calls via those numbers to return calls etc,
then you will probably need to add a prefix as the call comes in (Hack the
${CALLERID(num)} on the way in to add the prefix so the call goes back out
the right way.)
Rob
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Re: [asterisk-users] Get sip response code
On Thu, May 17, 2007 at 12:11:48AM -0600, Ken Williams wrote:
It's funny Robert would come looking for this tonight, as I've been
spending a fair amount of time trying to track this down today. I then
went to the source and found what Andreas had found below.
However, I'm not a real programmer, but just a hack of a hackI tried
to make my own variable but failed because I don't really know what the
hell I'm doing! :D
Here's one form of what I tried, though I did try lots of different ways,
but wasn't able to get it to compile without errors. At best I got the
server to do nothing, at worse I crashed the server when trying to use it:
I asked the question on digium bugs, and I got back a response along the
lines of: use ${HANGUPCAUSE}. They were not receptive to the idea of having
a SIP response code variable, or willing to discuss it, or the fact that my
original problem stems from the fact that CONGESTION is used for too many
things, not just CONGESTION, so it makes it difficult. It should really have
a FAIL response. (or just rename CONGESTION to FAIL since that's what it
acually means.)
It does seem strange though that you can see every sip header with
${SIP_HEADER(header)} but not the actul SIP response.
Hangupcause returns a value (including SIP channels) which is interpreted
back into a cause code RFC3398.
I have since update the wiki docs, as this was all a bit non-obvious to me:
http://www.voip-info.org/wiki/view/Asterisk+Variable+DIALSTATUS
http://www.voip-info.org/wiki/view/Asterisk+Variable+HANGUPCAUSE
Rob
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Re: [asterisk-users] SIP INVITE failing and AgentCallBackLogin()
On Wed, May 16, 2007 at 06:21:30AM -0700, Ron McCarthy wrote: Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to 'Ex 301 sip:[EMAIL PROTECTED] ;tag=as4e18cbb4' I have a peer setup in the box doing the AgentCallBackLogin() with insecure=very, ive also tried insecure=invite as well, no luck!! I'm not sure what the link you have here between SIP and Agents? Agents use chan_agent Dial(Agent/nnn) but SIP calls use chan_sip, so the two don't interact in the dialplan. (SIP User 301 is not equivalent to Agent 301, they are completely separate.) The error you have pasted here looks like either type= problem or the extension 301 doesn't exist in sip.conf of the box that the invite is being placed to. (or the IP address for the peer is wrong, etc.) I would not rely on AgentCallBackLogin(). chan_agent has limited use, which introduces a few strange problems, unchangable assumptions about how you want to handle calls, and the AgentCallBackLogin() feature has been (annoyingly) been deprecated by digium as of 1.4 The suggestion is to replicate the AgentCallBackLogin() functionality with dialplan logic, and dynamic queue members. This is possible, but very complicated (you do NOT want to see my extensions.conf!) and there is no neat way to handle hints for blf keys when you do this, as you lose the ability to dynamically track Agents in the hints config, and I haven't found a way to dynamically update the hints that doesn't crash asterisk. If you don't want BLF keys, this won't cause a problem. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip: [EMAIL PROTECTED] - inoc-dba: 5459*710- tel: +44 (0)20 7645 3510 ___ --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] Get sip response code
I was wondering if it is possible (in 1.2.x) to get the SIP response code back after doing Dial(). Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and some are NOANSWER, but I want to know the SIP response code, so I could return the right tones to the user, not just a congestion tone for every fault. Anyone know a way to find out that information, so I want the 604 out of this lot: -- Called [EMAIL PROTECTED] -- Got SIP response 604 Does Not Exist Anywhere back from x.x.x.x == No one is available to answer at this time (1:0/0/0) -- Executing NoOp(SIP/42105-d313f470, -- DIALSTATUS is: NOANSWER) in new stack -- Executing Goto(SIP/42105-d313f470, s-NOANSWER|1) in new stack -- Executing PlayTones(SIP/42105-d313f470, Unobtainable) in new stack -- Executing Wait(SIP/42105-d313f470, 10) in new stack Or where do I need to look to find a SIP response code - DIALSTATUS mapping? Are these configurable anywhere or are they hardcoded? If I push the response code back to the handset (Cisco 7960) then it is even more unhelpful as it uses the same error message for all SIP error type response codes: Reorder but does not tell you why the call failed to set up. If it actually put the SIP response error on the display, that would be good, but it doesn't. (at least SIP 8.6 and prior software versions) If it returns Congestion for, say, an invalid number destination, users hear what sounds like an engaged tone, when what I want them to hear is an unobtainable tone. Rob ___ --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] Get sip response code
On Wed, May 16, 2007 at 05:46:21PM -0500, Greg Oliver wrote: If I push the response code back to the handset (Cisco 7960) then it is even more unhelpful as it uses the same error message for all SIP error type response codes: Reorder but does not tell you why the call failed to set up. If it actually put the SIP response error on the display, that would be good, but it doesn't. (at least SIP 8.6 and prior software versions) In order to display the response on the handset, Cisco phones require that you have privileges and CTI control over the phones. The only un-authenticated things you can display through the phones are through the services and directories keys. Unfortunately, they are keeping that locked up since they want you to use them with their system. Hopefully they will change their minds one day. Yes. I know that... This is exactly the limitation I am trying to work around, by being able to send back meaningful tones to the phone from asterisk in-band rather than sending back the SIP response codes which all get displayed by the handset as Reorder which is completely useless in informing the user what's wrong. (And the US reorder tone sounds too much like the UK engaged tone anyway.) Even if the handset did display the SIP error response, I'm not expecting most users to understand the subtleties of what they all mean, so it seems better just to simplify it with a few well known tones most users are already familiar with (unobtainable, equipment busy, user busy, etc.) And it will behave in the same way regardless of the model of handset. (Call worked/Busy/Call failed...) Unfortunately Dial() DIALSTATUS is a bit limited in that if call setup fails for some reason, it mostly returns CONGESTION. Playing a congestion tone for perhaps 12 different call setup problems including misdials, doesn't help either. I want to play the right tone (for, say, unobtainable, equipment busy, etc.) The ISDN gateway I am using goes to great pains to send back the correct SIP response to asterisk, which then just reports it as CONGESTION which is a bit limiting. The SIP response code is displayed on asterisk's console, I just cannot see a way to get at it in the dial plan Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan / problem with extension-length 1
On Wed, Apr 25, 2007 at 01:21:40PM +0200, Michael Kamleitner wrote:
hi community,
I'm new to this list asterisk in general, so let me first say thx to
everybody involved in providing such great tools ressources!!
I'm currently trying to implement a simple voicebox-system.
for demonstration purposes, I've successfully connected my cellphone via
bluetooth using the current chan_cellphone-patch on the current SVN-version
of asterisk. everything seems to work fine so far (great patch!)
what I want to achieve:
* incoming call arrives
* asterisk/cellphone answers
* caller is greeted (playback of my-intro)
* caller enters an extension
* caller is directly forwarded to the voicemail of entered extension
I think waitexten is only for getting one (optional) digit at a time,
for building IVR menus and things like that.
If the thing you want entering is non-optional or more than one digit,
you may be better off using the Read() command. See:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Example of:
exten = start,1,Read(agent,agent-user);
Plays the sound file please enter your agent number, followed by the hash
key and puts the result into the variable ${agent}
You can also set the maximum number of digits to read, and a timeout, etc.
Rob
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Re: [asterisk-users] Asterisk dialing next extension only if first is busy?
On Mon, Apr 23, 2007 at 11:11:48AM -0500, Carlos Chavez wrote: Using two sequential Dial() commands into the extension will ring the lines one after the other -- even if it times out on the first line, which is again not what I want. I find that the easiest way to do it is like this: 1,1,Dial(SIP/line1) 1,2,Dial(SIP/line2) Than way if the first like fails for any reason it goes to the second. You could use Dialstatus but this seems simpler. Not necessarily. If the handsets have call waiting or divert enabled for example it will go to the first dial instance and not fail through to the second. This may or may not be the desired behaviour depending on what you want to happen, of course. Rob ___ --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] Internet gateway problem
On Mon, Apr 23, 2007 at 11:12:15AM +0200, voip crazy wrote: Hello all, I have got an asterisk server in my LAN, getting access to internet trought a router. I have observed in my asterisk box, when the internet connection in down, the phones can not register to my asterisk. It is like chan_sip, does not work without an internet connection. If when the router is down the telephones does not register, but when I type in my asterisk box route del default, teh phones then started to register against the asterisk. Is there any debugging information available from the logs/console? Does your server have a fixed IP address or does it change about? Why this is happenning? Why chan_sip, need a gateway or it does not start correctly? Why when I type route del default the phones started to register? I have seen similar issues recently with asterisk SIP service when DNS becomes unavailable, the chan_sip dies because the DNS lookups are blocking, and it has to wait for every request to time out. If your box has no default route, then it will respond immediately with a Network Unreachable message rather than wait for the DNS to time out. If you have any outbound register = entries try taking them out and experimenting with the line down. Also if you have any hints in extensions.conf try taking those out. Asterisk seems to make many spurious DNS requests. http://bugs.digium.com/view.php?id=9536 A possible workaround which I have implemented here, is to have a local instance of BIND on the asterisk box which slaves for the local zones and caches things, so that it does not die so often. Rob -- Robert Lister - London Internet Exchange- http://www.linx.net/ [EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE ___ --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 dialing next extension only if first is busy?
On Mon, Apr 23, 2007 at 06:18:32PM +1000, Daniel Pittman wrote: G'day. I am having reasonable success getting Asterisk 1.4.2 running and doing what I want, but I can't figure out one particular idiom that I want: There are a few situations where I want to have Asterisk push a call through to the first available transport on a list, such as: I have two SIP ports attached to one local (two port) analog phone system. I want to ring line 1 for the first call, line 2 for the second call and go to voicemail for the third and subsequent. I can't work out the best way to express that. Using Dial(SIP/line1SIP/line2) will ring both lines at the same time which is not really what I want. You might want to look at doing this with a queue, and then directing the call into the queue. There are some new queue strategies in 1.4.x that might do what you want, and it also has autofill option which might make it behave the way you want. There is also a linear type strategy which looks like it is making its way into the code, which might be more suitable than roundrobin/rrmemory. http://bugs.digium.com/view.php?id=7279 Or, you might be able to implement it by using the ChanIsAvail command in the dialplan (If the device is returning reasonable things.) It can be used to test availability of a channel or a list of channels and returns the status, or the available channel name. I do a similar thing here and it works very well. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail Rob -- Robert Lister - London Internet Exchange- http://www.linx.net/ [EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE ___ --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] G.729 Voicemail
On Fri, Apr 20, 2007 at 02:59:28PM +0200, Michael Landin Hostbaek wrote: List, I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication between the phones is G.729, and my sip.conf looks like this: disallow=all; First disallow all codecs allow=g729 ; allow=gsm allow=ulaw allow=alaw However, I cannot call voicemail - I get the following error: [Apr 20 14:58:31] WARNING[87184]: channel.c:2816 set_format: Unable to find a codec translation path from g729 to gsm Shouldn't it switch to gsm automatically? Cisco 79XX phones only support ulaw, alaw or g729, not gsm. Asterisk only supports g.729 protocol in passthrough mode without the licence (i.e. It can set up a session between two licenced g.729 endpoints to talk to each other, but cannot get into the media path itself.) The voicemail system is presumably trying to transcode from g.729 to gsm and you haven't got the licence for that. (Maybe you can get hold of/convert the sounds in the g729 format for the voicemail system, then it may not have to transcode out of .gsm?) I am not sure what parts of the system are enabled/disabled without the licence. http://www.voip-info.org/wiki-Asterisk+G.729+Licensing I cannot purchase g729 licenses, as FreeBSD is not yet supported (with asterisk 1.4) ___ --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] G.729 Voicemail
transcode out of .gsm?) I am not sure what parts of the system are enabled/disabled without the licence. This mentions voicemail g729 in pass-thru mode. I'm not sure if it works as I've never tried it, but it may be worth a try... http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru -- Robert Lister - London Internet Exchange- http://www.linx.net/ [EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE ___ --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] Best External PRI Gateway?
On Thu, Apr 12, 2007 at 11:59:00AM -0400, jameson asterisk wrote: I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 Also have a look at Patton SmartNode 4960 range. They are available in various configurations/numbers of channels, some of which are upgradable to more channels at a later date: http://www.patton.com/products/pe_printable.asp?category=354 We have the ISDN2 and Analogue versions of these gateways (same software) and so far they have been very reliable, and can be configured in a variety of fail-over situations in case asterisk or the connection to the server dies, incoming calls can be automatically routed either back out on another ISDN channel or out to another SIP/analogue gateway etc. Rob -- Robert Lister - London Internet Exchange- http://www.linx.net/ [EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to return dialstatus of second (sub) call
On Thu, Apr 05, 2007 at 02:06:53AM -0500, Jonathan Rivera wrote:
Hello all
I have this problem, i need a way to balance my trunks which are SIP
peers, when a SIP peer is busy then send the call for another peer and
so until i can send away the call, i think i can do it with queues.
Ok this is the scenario:
In extensions.conf
[balance]
exten = _,1,NoOp(Call to: ${EXTEN})
exten = _,2,Answer()
exten = _,3,SetVar(_ORGEXTEN=${EXTEN})
exten = _,4,SetVar(_ORGUNIQUEID=${UNIQUEID})
exten = _,5,Set(CDR(userfield)=${ORGUNIQUEID})
exten = _,6,Queue(qtest,r)
exten = _,7,Hangup()
I have a queue with 100 members which are local channels
In queues.conf
[qtest]
strategy=random
member=Local/[EMAIL PROTECTED]
member=Local/[EMAIL PROTECTED]
member=Local/[EMAIL PROTECTED]
I had a similar problem of returning state to the queue manager to check the
call state.
You might want to try something like:
exten = check,1,ChanIsAvail(Local/[EMAIL PROTECTED],js);
exten = check,102,Goto(busy,1);
exten = busy,1,Busy();
Obviously you could replace this with a macro/DB lookup to avoid having lots
of repeated entries in the dial plan.
Busy() should return busy to the queue application if the Local
channel is in use, causing it to skip to the next entry in the queue.
After having a nightmare with chan_agent not working properly, I implemented
a modified (for 1.2.x) version of:
http://www.voip-info.org/wiki/view/Agents+without+agent+channel
and stopped using AgentCallBackLogin(), which digium it appears have
deprecated anyway in 1.4.x
Agents without agent channel is a bit of a hack, but it works better than
chan_agent in my case.
This caused various other problems, notably that hints do not seem to work
with Local/ channels, it shows them as always available. I have not found
a workaround to this as yet. Any attempts I have made to dynamically update
hints in the dialplan from asterisk CLI (add extension .) seems to
cause it to core dump in my case. Other than that, it works quite well.
Rob
--
Robert Lister - London Internet Exchange- http://www.linx.net/
[EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE
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Re: [asterisk-users] Using server side phonebook directory with SPA941
On Tue, Mar 27, 2007 at 12:45:44PM +0200, Maxim Veksler wrote:
Hello list,
I got a couple of those wouldn't it be great questions, as following:
1. Is it possible, with asterisk to hold a central phonebook directory
of callers?, so that when this party calls a textual caller ID will
be displayed on the phone display.
Can be done reasonably easily in the dial plan. What I have is quite noddy
but it does the job. In the incoming bits of dial plan where calls come in,
I call this as a macro in the context where incoming calls arrive, before
handing it off to the Dial() bits:
exten = _4535XX,1,Macro(setisdncallerid,${EXTEN},PSTN,9)
What this macro (pasted below) does is allow alpha tagging of incoming
calls, plus some defaulty stuff set by the gateway (caller ID not
present/withheld comes through in my case as either anonymous or just 0 or
00, so this macro tidies this up before passing the call on.)
It also inserts the access digit (9) in front of the caller ID as in my case
outside calls need a 9 prefix. This is just so that call routing works
correctly if people return missed calls/save numbers from the handset etc.
Obviously you will have to tweak this for your setup.
If there is no alpha tag in the DB, it sets some defaulty thing (In my case
PSTN to give some indication where the call is coming from.)
It can also do a CPI tag based on destination number, for queues/group
numbers, so that the alpha tag on the call gets set to something like
Main Number etc. to distinguish a DDI call from a Queue Call.
The database entries look like:
*CLIdatabase put tag 01234567890 Some Name Here
and for CPI (called party) Tag:
*CLIdatabase put 453510 tag Helpdesk
[macro-setisdncallerid]
; ${ARG1} = Called Party Number (XX) as presented from BT.
; ${ARG2} = default tag to add to incoming calls
; ${ARG3} = prefix to insert to incoming CLI
;
; Frobs the incoming caller ID headers how we like it:
exten = s,1,NoOp(macro-setisdncallerid: ${ARG1})
; In my case the internal extension is 7XX where XX is the
; last two digits of the incoming DDI number. This just makes
; it display right in the caller ID:
exten = s,2,Set(DIALED_EXTEN=7${ARG1:-2})
; For cisco phone, set different ring cadence to indicate
; an external call:
exten = s,3,SIPAddHeader(Alert-Info: Bellcore-dr2)
exten = s,4,GotoIf($[ ${CALLERID(num)} = anonymous ]?400)
exten = s,5,GotoIf($[ ${CALLERID(num)} = 0 ]?500)
exten = s,6,GotoIf($[ ${CALLERID(num)} = 00 ]?500)
exten = s,7,GotoIf($[ ${DB(tag/${CALLERID(num)})} != ]?700)
exten = s,8,Set(CALLERID(name)=${ARG2} to ${DIALED_EXTEN})
exten = s,9,Set(CALLERID(num)=${ARG3}${CALLERID(num)})
exten = s,10,Goto(900)
exten = s,400,Set(CALLERID(name)=${ARG2})
exten = s,401,Goto(900)
exten = s,500,Set(CALLERID(num)=unknown)
exten = s,501,Set(CALLERID(name)=${ARG2})
exten = s,502,Goto(900)
exten = s,700,Set(CALLERID(name)=${DB(tag/${CALLERID(num)})})
exten = s,701,Set(CALLERID(num)=${ARG3}${CALLERID(num)})
exten = s,702,Goto(900)
; If there is a CPI tag set, use that: (i.e. SUPPORT)
exten = s,900,GotoIf($[ ${DB(${ARG1}/cpitag)} != ]?950)
exten = s,950,Set(CALLERID(name)=${DB(${ARG1}/cpitag)})
2. How can this be configured with Trixbox, I've looked at the
configuration options - I assume it plays no difference me basing it
on mysql or astdb?
3. What protocol does the phone (Linksys SPA941) talks to the
asterisk server to retrieve this information ?
When an incoming call arrives with asterisk, the SIP headers can be set
appropriately before you present this information to the handset. It's in
the incoming SIP packets to the handset.
4. Has someone done this? What softphone should I use to test it first
(I'm connecting it with outlook, so it has to be win* software)
There are a few to choose from. I use Counterpath's X-Lite client:
http://www.counterpath.com/
Rob
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Re: [asterisk-users] Limit call duration
On Wed, Mar 21, 2007 at 12:56:55PM +0100, Suity Zsolt wrote: Hi everyone, I'm new to Asterisk, but I like it ;o) Have a question to you; How can I limit the incoming call duration? I think you can say something like: AbsoluteTimeout (or in 1.2x, Set(TIMEOUT(absolute) = seconds) ) See: http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout Rob ___ --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] automated dialout detect forward
On Wed, Mar 21, 2007 at 01:23:37PM +0100, Mike Heininger wrote: Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. One simple method would be to dial out and then playback an announcement announcing the incoming call, maybe even the number, and ask the user to press some key to accept the call. If this key is not pressed within a certain timeout, then terminate. This is okay to detect answering machines etc. I believe asterisk 1.4. has some better controls over this. In 1.2, some other techniques are discussed at: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackgroundDetect Rob -- Robert Lister - London Internet Exchange- http://www.linx.net/ [EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE ___ --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] FWD outgoing problem
On Wed, Mar 21, 2007 at 04:40:02PM +0200, Bogdan Gonciulea wrote:
[globals]
FWDNUMBER=yy
FWDPASSWORD=
FWDCIDNAME=some name
[default]
exten = _393.,1,Set(CALLERID(all)=${FWDCIDNAME})
exten = _393.,n,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
exten = _393.,n,Congestion
I have also took out the Set(CALLERID...) line and the result was the same.
That doesn't look entirely right to me, maybe it should be:
Set(CALLERID(name)=${FWDCIDNAME})
Set(CALLERID(num)=${FWDNUMBER})
Set(CALLERID(all)= is for setting the entire caller ID header, so it
should look something like this if you use it:
Set(CALLERID(all)=Joe User 1234)
I think the things after DIAL(IAX2/..) should match what you have configured
in iax.conf for iax peer:
iax.conf (from some example I found):
[FWDIAXPeer]
type=peer
disallow = all
allow=ulaw ; FWD only do ulaw
host=iax2.fwdnet.net
qualify=300 ; optional of course
secret=secret
context=from-fwd
username=321321
Then: Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:3},45)
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