Re: [asterisk-users] is roundrobin and rrmemory the same meaning?
In article a160a7d6100926h6d2e6f88m64175b92cfcc2...@mail.gmail.com, Zhang Shukun bit...@gmail.com 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? ; A strategy may be specified. Valid strategies include: ; ; ringall - ring all available channels until one answers (default) ; roundrobin - take turns ringing each available interface ; leastrecent - ring interface which was least recently called by this queue ; fewestcalls - ring the one with fewest completed calls from this queue ; random - ring random interface ; rrmemory - round robin with memory, remember where we left off last ring pass ; ;strategy = ringall Both roundrobin and rrmemory will ring phones one at a time, for the length of time given in timeout, and then if not answered will move along to the next phone and ring it. Let's say you have three of more phones in the queue. Phone 1 gets rung but not answered, then Phone 2 gets rung and is answered. When another call comes in, roundrobin would start again with Phone 1, but rrmemory would start with Phone 3, as it was Phone 2 that picked up the last call. Hope this helps, Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is roundrobin and rrmemory the same meaning?
Thank you! it's very helpful. now i have another question: in asterisk, each agent should login first and then can response to the caller. but i don't want to the login action. i need agent shold response directly without login first. how should i do ? can users in sip.conf to be agents? so it can login persistently on a phone. 2010/1/12 Tony Mountifield t...@softins.clara.co.uk: In article a160a7d6100926h6d2e6f88m64175b92cfcc2...@mail.gmail.com, Zhang Shukun bit...@gmail.com 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? ; A strategy may be specified. Valid strategies include: ; ; ringall - ring all available channels until one answers (default) ; roundrobin - take turns ringing each available interface ; leastrecent - ring interface which was least recently called by this queue ; fewestcalls - ring the one with fewest completed calls from this queue ; random - ring random interface ; rrmemory - round robin with memory, remember where we left off last ring pass ; ;strategy = ringall Both roundrobin and rrmemory will ring phones one at a time, for the length of time given in timeout, and then if not answered will move along to the next phone and ring it. Let's say you have three of more phones in the queue. Phone 1 gets rung but not answered, then Phone 2 gets rung and is answered. When another call comes in, roundrobin would start again with Phone 1, but rrmemory would start with Phone 3, as it was Phone 2 that picked up the last call. Hope this helps, Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- 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?
Zhang Shukun wrote: Thank you! it's very helpful. now i have another question: in asterisk, each agent should login first and then can response to the caller. but i don't want to the login action. i need agent shold response directly without login first. how should i do ? can users in sip.conf to be agents? so it can login persistently on a phone. My phones are listed in queues.conf member = SIP/36949608 member = IAX2/10 member = IAX2/11 -- _ -- 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] Using HASH() and REALTIME_HASH()
Le 10/01/2010 07:53, Tilghman Lesher a écrit :
On Saturday 09 January 2010 15:22:29 Benoit wrote:
I'm playing around with asterisk 1.6.2.0 and the first try was to
replace my now non-functionning
'app-realtime' macro which emulated RealTime with REALTIME_HASH()
There is very few documentation on the subject except for this bug report:
https://issues.asterisk.org/view.php?id=13651#c94998
However when i try this syntax:
Set(HASH(info)=${REALTIME_HASH(call_info,exten,${dest})});
the syntax doesn't seem to be happy:
-- Executing [...@appel_deb:8] Set(SIP/maverick-,
HASH(info)=,101,maverick,0,0,max,0,0,123456,123654) in new stack
[Jan 9 22:07:25] WARNING[27801]: pbx.c:9107
pbx_builtin_setvar_multiple: MSet: ignoring entry '101' with no '=' (in
s...@appel_deb:8
[Jan 9 22:07:25] WARNING[27801]: pbx.c:9107
pbx_builtin_setvar_multiple: MSet: ignoring entry 'maverick' with no '='
(in s...@appel_deb:8
[Jan 9 22:07:25] WARNING[27801]: pbx.c:9107
pbx_builtin_setvar_multiple: MSet: ignoring entry '0' with no '=' (in
s...@appel_deb:8
[Jan 9 22:07:25] WARNING[27801]: pbx.c:9107
pbx_builtin_setvar_multiple: MSet: ignoring entry '0' with no '=' (in
s...@appel_deb:8
I had to do the following:
Set(HASH(info)=${REALTIME_HASH(call_info,exten,${dest})});(adding
of double quote)
Yes, this is because you're on a machine that you upgraded from 1.4. This
makes Set get the old 1.4 behavior that I tried to leave behind. In your
asterisk.conf file, create or modify the following section:
[compat]
app_set=1.6
and it will start working beautifully, in an intuitively obvious way.
Hi,
Thank you it does indeed fix the problem, i should have read more
carefully the UPGRADE-1.6.txt before posting :(
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make the first one work,
using DAHDI or mISDN
under asterisk 1.6.
Asterisk does receive inbound calls, with extensions informations and
all but when going to the point
of actually dialing a phone and connect it to the call it look like
stuck, well not totally stuck since the
Dial's timeout is working and all but the sip phone isn't ringing,
asterisk isn't reporting that the phone
is ringing and the call end's up to voicemail which is working at least
for emitting audio, i have not tested
recording
Calling through the TE220 (working):
-- Executing [...@appel_deb:46] Dial(DAHDI/1-1, SIP/benoit,8,tTwW,)
in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called benoit
-- SIP/benoit-0032 is ringing
-- Channel 0/1, span 1 got hangup request, cause 16
Calling through the B410p (not working):
-- Executing [...@appel_deb:46] Dial(DAHDI/63-1,
SIP/benoit,8,tTwW,) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called benoit
-- Channel 0/1, span 3 got hangup request, cause 16
Any idea ?
regards
--
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[asterisk-users] Beginners Guide to setting up a Call Centre
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. Peter. -- _ -- 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
http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- 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] Virtual ISDN device /dev/XYZ
Hello, I do remember having read some weeks ago something about a virtual device provided by asterisk, behaving like an ISDN device, i.e. like /dev/isdn0. I know iaxmodem, but iaxmodem imho unfortunately does not transport raw ISDN data (HDLC frames), but only voice. Do I remember right, and there is an aseterisk application, providing such a device, which other linux executables can use, which expect a common ISDN device? Roger. -- _ -- 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
2010/1/12 James Mutuku listmut...@gmail.com: http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a I can use Google just as well as the next guy, and if you'd bothered to look at the results you could see they were extremely bland and not partially useful. I'm thinking I want some up to date information and a beginners guide, But I'm finding it difficult to find much dated after 2003 I'm not an expert on phones, I'm just an IT guy who thinks he might have a solution to a problem, that is not really his problem but is trying to see if he can get it to work. That's how bad the Alcatel phone system is! Peter. -- _ -- 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
Since you are small, trixbox would probably be the ideal flavor of Asterisk for you. It is a downloadable ISO that installs Scientific Linux and Asterisk and sets you up to manage everything with a GUI interface from a browser. Once you outgrow that, you can either expand it, go for Commercial Asterisk or join the fun world of Open Source Asterisk where we work on releases and/or SVN branches. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Childs Sent: Tuesday, January 12, 2010 4:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Beginners Guide to setting up a Call Centre 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. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
Peter Childs wrote: I'm not an expert on phones, I'm just an IT guy who thinks he might have a solution to a problem, that is not really his problem but is Then you'll need to be prepared to do a LOT of reading. You'll want to start off on: http://voip-info.org Then there is the Asterisk documentation project: http://www.asteriskdocs.org/ You'll also want to download and read the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf Doug -- _ -- 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
On Tue, 12 Jan 2010, Danny Nicholas wrote: Since you are small, trixbox would probably be the ideal flavor of Asterisk for you. It is a downloadable ISO that installs Scientific Linux and Asterisk and sets you up to manage everything with a GUI interface from a browser. Once you outgrow that, you can either expand it, go for Commercial Asterisk or join the fun world of Open Source Asterisk where we work on releases and/or SVN branches. I agree that FreePBX would be the ideal flavor for him, but I am a recent convert to Elastix. Much tighter GUI, more included stuff (like hylafax and iaxmodem), and just overall a better stab at the whole integration. After two horrid experiences with Trixbox Pro and my impression of Elastix over Trixbox CE I will never install another Fonality product. j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Childs Sent: Tuesday, January 12, 2010 4:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Beginners Guide to setting up a Call Centre 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. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
I actually meant switchvox (just to make the content of my comment be kosher), but in general, the OP should probably go with a canned solution unless he wishes to get his hands dirty. -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, January 12, 2010 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Beginners Guide to setting up a Call Centre On Tue, 12 Jan 2010, Danny Nicholas wrote: Since you are small, trixbox would probably be the ideal flavor of Asterisk for you. It is a downloadable ISO that installs Scientific Linux and Asterisk and sets you up to manage everything with a GUI interface from a browser. Once you outgrow that, you can either expand it, go for Commercial Asterisk or join the fun world of Open Source Asterisk where we work on releases and/or SVN branches. I agree that FreePBX would be the ideal flavor for him, but I am a recent convert to Elastix. Much tighter GUI, more included stuff (like hylafax and iaxmodem), and just overall a better stab at the whole integration. After two horrid experiences with Trixbox Pro and my impression of Elastix over Trixbox CE I will never install another Fonality product. j -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Childs Sent: Tuesday, January 12, 2010 4:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Beginners Guide to setting up a Call Centre 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. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
Jeff LaCoursiere wrote: I agree that FreePBX would be the ideal flavor for him, but I am a recent convert to Elastix. Much tighter GUI, more included stuff (like And, I'd be in the camp that would advocate getting your hands dirty and learn to program without the GUI. You'll learn a lot and then if you'd want to move to a GUI and something breaks, you'll have some idea on what and how to fix it. Knowing now what I do, I find a GUI to restrictive. Doug -- _ -- 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
And, I'd be in the camp that would advocate getting your hands dirty and learn to program without the GUI. You'll learn a lot and then if you'd want to move to a GUI and something breaks, you'll have some idea on what and how to fix it. Knowing now what I do, I find a GUI to restrictive. I agree. I originally felt I wanted the GUI approach too, but then when I looked into things in more detail and understood that you really can't BOTH use the GUI approach and edit files explicitly, I decided that the GUI did nothing for me except add a additional level of complexity and that I'd be MUCH better off just doing things directly. -- _ -- 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] Hardware issue, was Using HASH() and REALTIME_HASH()
On Tuesday 12 January 2010 04:44:36 Benoit wrote: I just experienced another problem however i have two rnis cards, one b410p and one te220, while the later works prefectly i can't really make the first one work, using DAHDI or mISDN under asterisk 1.6. If you're having trouble with any Digium hardware, the best thing to do is to call Digium support and get your free installation support provided with our hardware. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net wrote: Codec? I've had 2833 do funny things with anything other than a/ulaw (might just be me though..) S -- Codecs other than G711u/a don't support inband DTMF. Seeing as INFO is rarely used that pretty much leaves RFC2833. Turn on SIP debugging and look in the INVITE from the provider for telephone-event. If you see it, they're configured to use RFC2833. If they are, you need to do a packet capture or other RTP debug to see the out of band RFC2833 events. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
Yes it is - we have thousands of happy clients worldwide. :) My suggestion is to go for somebody who has relevant experience and is going to do the install for you. Unless your CC is very small, you don't want to be looking up the manuals when you went live and start having quality issues If you need some pointers in your area, please contact me off-list. l. 2010/1/12 Peter Childs pchi...@bcs.org 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. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime LDAP Queues crashes
On Friday 08 January 2010 01:38:42 Gavin Henry wrote: What are the LDAP searches like? after updating and applying this patch: http://issues.asterisk.org/view.php?id=13573 doesn't crash and the queries i get are ok: conn=0 op=67 SRCH base=dc=nodomain scope=2 deref=0 filter=((objectClass=AsteriskQueue)(AstQueueName=barbaros)) = bdb_equality_candidates: (AstQueueName) not indexed conn=0 op=67 ENTRY dn=cn=barbaros,ou=queues,dc=nodomain conn=0 op=67 SEARCH RESULT tag=101 err=0 nentries=1 text= conn=0 op=68 SRCH base=dc=nodomain scope=2 deref=0 filter=((objectClass=AsteriskQueueMember)(AstQueueInterface=*) (AstQueueMemberof=barbaros)) = bdb_equality_candidates: (AstQueueMemberof) not indexed conn=0 op=68 ENTRY dn=uid=1234,ou=users,dc=nodomain conn=0 op=68 ENTRY dn=uid=demo,ou=users,dc=nodomain conn=0 op=68 SEARCH RESULT tag=101 err=0 nentries=2 text= but the queue is shown as empty: -- Executing [...@users:1] Queue(SIP/jsalamero-0001, barbaros) in new stack [Jan 12 16:32:37] WARNING[4238]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:32:37] WARNING[4238]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo -- Started music on hold, class 'default', on channel 'SIP/jsalamero-0001' voip*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Realtime 1234/1234 87.222.XXX.XXX D N 5060 OK (91 ms) Cached RT jsalamero/jsalamero87.222.XXX.XXX D N 1024 OK (86 ms) Cached RT /94.23.xxx.xxx5060 Unmonitored 3 sip peers [Monitored: 2 online, 0 offline Unmonitored: 1 online, 0 offline] voip*CLI queue show barbaros barbaros has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s No Members Callers: 1. SIP/jsalamero-0001 (wait: 0:44, prio: 0) [Jan 12 16:33:21] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:33:21] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo [Jan 12 16:33:21] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:33:21] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo after adding by hand the users 1234 and demo to the queue, it works: queue add member SIP/demo to barbaros queue add member SIP/1234 to barbaros voip*CLI queue show barbaros barbaros has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:2, SL:0.0% within 0s Members: SIP/demo (dynamic) (Not in use) has taken no calls yet SIP/1234 (dynamic) (Not in use) has taken no calls yet No Callers voip*CLI [Jan 12 16:42:24] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:42:24] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo [Jan 12 16:42:24] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:42:24] WARNING[4227]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo -- Executing [...@users:1] Queue(SIP/jsalamero-0005, barbaros) in new stack [Jan 12 16:42:51] WARNING[4754]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber 1234 [Jan 12 16:42:51] WARNING[4754]: app_queue.c:1855 rt_handle_member_record: Realtime field uniqueid is empty for memeber demo -- Started music on hold, class 'default', on channel 'SIP/jsalamero-0005' -- SIP/demo-0007 is ringing -- SIP/1234-0006 is ringing -- Stopped music on hold on SIP/jsalamero-0005 == Spawn
Re: [asterisk-users] is roundrobin and rrmemory the same meaning?
You can list phones directly as static members of the queue. this is generally sub.optimal because if. e.g. an agent of yours is home sick, her phone will be ringing and you'll be wasting caller time. Also by tracking logins and logoffs you can measure agent productivity, and this is pretty useful in most environments. l. 2010/1/12 Zhang Shukun bit...@gmail.com Thank you! it's very helpful. now i have another question: in asterisk, each agent should login first and then can response to the caller. but i don't want to the login action. i need agent shold response directly without login first. how should i do ? can users in sip.conf to be agents? so it can login persistently on a phone. -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
On Tue, 12 Jan 2010, Richard Kenner wrote: And, I'd be in the camp that would advocate getting your hands dirty and learn to program without the GUI. You'll learn a lot and then if you'd want to move to a GUI and something breaks, you'll have some idea on what and how to fix it. Knowing now what I do, I find a GUI to restrictive. I agree. I originally felt I wanted the GUI approach too, but then when I looked into things in more detail and understood that you really can't BOTH use the GUI approach and edit files explicitly, I decided that the GUI did nothing for me except add a additional level of complexity and that I'd be MUCH better off just doing things directly. That is so not true. FreePBX has hooks in a million places to do custom dialplan stuff - I do it all the time. I also link in custom AGI/AMI applications, custom provisioning, custom LCR, and am even working with one customer that has mastered making FreePBX multi-tenant. If you want to get your hands dirty there is plenty of dirt underneath FreePBX. On the other hand, if you want a simple setup that is easily managed, the GUI is fantastic and saves a LOT of time. And if you are a PHP programmer you can easily modify the operation of any part of it. Your comments both come from having taken a short look at FreePBX and dismissed it without investigating how powerful it can be. Now as far as Switchvox goes, now THAT is a restrictive platform. You cannot ssh into the box for starters. Every extension requires a license. There is no support for dual homing the box (my default installation configuration - one port on public!). Another horrid experience. j -- _ -- 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
Your comments both come from having taken a short look at FreePBX and dismissed it without investigating how powerful it can be. Yes, but the discussion is about COMPLEXITY, not power! Sure, there are hooks where you can do anything you want, but if you were to set up identical configurations via FreePBX and by writing a dialplan (and other config files) from scratch, the latter will be the least complex. What that means is that if your goal is to learn the least about Asterisk that you can get away with, but that you expect to need to tweak the dialplan, doing so is going to have a lower learning curve if you JUST use Asterisk: using FreePBX just means that you have to learn BOTH systems and that you'll be modifying a more complex configuration than if you did it yourself. -- _ -- 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] Send 503 or 603 error after answer()
Hello list. Is it possible in the Asterisk dialplan to send a 503 Service Unavailable of 603 Decline after having answered the call with Answer() in the dialplan ?? Suppose that I first want to check the call in a MySQL-database, while I put some MoH, and then let the call go through or send some error to my ITSP where the call comes from. I know there is something like 'early media', but isn't it good practise to always have Answer() in the dialplan ? Another thing why I want to have this 503 or 603 after having Answered the call : my ITSP offers the ability to have a backup number for the call if it can't get through. If I send the call to a queue and there is nobody in the queue at that moment, I want to send a sort of error so my ITSP will send the call to a backup number, like my cellphone. If I send the call myself to my cellphone, I will have to pay big time and it's a waist of bandwidth... Kind regards, Jonas. -- _ -- 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] Sipgate DTMF not detected
On Tue, 2010-01-12 at 10:37 -0500, Kristian Kielhofner wrote:
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net
wrote:
Codec? I've had 2833 do funny things with anything other than a/ulaw
(might just be me though..)
S
--
Codecs other than G711u/a don't support inband DTMF. Seeing as INFO
is rarely used that pretty much leaves RFC2833. Turn on SIP debugging
and look in the INVITE from the provider for telephone-event. If you
see it, they're configured to use RFC2833.
If they are, you need to do a packet capture or other RTP debug to see
the out of band RFC2833 events.
--
Kristian Kielhofner
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
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[asterisk-users] Question about SIP registration
Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) Then I have configured an account as following: [999] type=friend username=999 host=dynamic port=5080 context=sipfrom nat=no canreinvite=no call-limit=8 videosupport=no disallow=all allow=alaw qualify=15000 So far, so good. Now, I have an internal process (onto Linux PC) which is a SIP endpoint and should register to Asterisk as 1.1.1.1:5080, but an external entity (i.e. a SIP endpoint over public Internet) is trying to register to Asterisk as 9...@89.x.y.zmailto:9...@89.x.y.z:5060 and the registration SUCCEEDS! When I launch the CLI command sip show peers, I see a row like this: 999/9991.1.1.1 5060 OK (3 ms) 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? Thanks for your help! Regards, Alberto Aggio -- _ -- 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
On Tue, 12 Jan 2010, Richard Kenner wrote: Your comments both come from having taken a short look at FreePBX and dismissed it without investigating how powerful it can be. Yes, but the discussion is about COMPLEXITY, not power! I thought the discussion was about how an IT guy with no previous asterisk experience could get up and running the fastest. By FAR that answer is to use one of the pre-packaged installations such as TrixBox or Elastix. Sure, there are hooks where you can do anything you want, but if you were to set up identical configurations via FreePBX and by writing a dialplan (and other config files) from scratch, the latter will be the least complex. By whose estimation? To even get that far with asterisk requires a lot of reading and experience. It took me several weeks to get my first installation answering the phone in 2003, before there were any serious GUIs available. My first intallation of aster...@home, however, was answering the phone in about 2 hours. What that means is that if your goal is to learn the least about Asterisk that you can get away with, but that you expect to need to tweak the dialplan, doing so is going to have a lower learning curve if you JUST use Asterisk: using FreePBX just means that you have to learn BOTH systems and that you'll be modifying a more complex configuration than if you did it yourself. The thing is the OP probably won't need to tweak the dialplan to do what he needs to do. My take is this - if you want to get started with Asterisk and you have NO experience, a pre-built package like Asterisk NOW, PIF, Trixbox, or Elastix is the quickest and cleanest way to get setup and running. After having some experience with it and finding the things that may require some custom dialplan work (getting harder and harder to find given the most recent releases of FreePBX and the things possible from the GUI), you can then learn the internals of dialplan coding and work that out over time. For someone starting from scratch, learning to setup Asterisk properly and coding your first diaplan - even using the samples - is difficult and non-intuitive. j -- _ -- 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] Inserting a wait in a sip dial
Hi All,
After searching and didnt found it, im just sending my situation here,
maybe someone knows where i should look.
Im using Asterisk 1.6.1.10
Internally the user with a sip phone dials a number for instance 0623456789
It goes fine to the specific dial rule:
which is: exten = _0[6].,2,Dial(SIP/0${EXTEN:1...@xs4all-out,60,tTwWkK)
This works fine without a charm, but the situation is that i want to hide
the phonenumber going out, this is done in the netherlands by dialling
*31# and then the phonenumber you want to call.
so i modified it to:
exten = _0[6].,2,Dial(SIP/*31#0${EXTEN:1...@xs4all-out,60,tTwWkK)
Only then it doesnt work, since i prolly need to wait before dialling the
number.
so after searching i saw several posts and sites which stated that i need
to use 'w' in the dial command.
So i changed it to:
exten = _0[6].,2,Dial(SIP/*31#w0${EXTEN:1...@xs4all-out,60,tTwWkK)
But then the other peer says:
-- Called *31#w06123456...@xs4all-out
-- SIP/xs4all-out-0234 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION'
Anyone an idea where i should look, or how i should change it, so that i
do get a wait before sending the rest of the number to the sip peer.
Thanks in advance,
Regards,
Evert
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[asterisk-users] SIP Security
Hey guys, I've been running asterisk on my server for some time now (currently running Asterisk 1.6.2.0). I am having security issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid=blah XX host=dynamic nat=yes canreinvite=no mailbox=1...@default registertrying=yes [testuser] type=friend secret=blah callerid=blah X host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default [testuser2] type=friend username=testuser2 secret= callerid=blah blah host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default Someone is able to connect to my server and make a call since they can access the default context. What should I do? Thanks guys! -- _ -- 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
Instead of host=dynamic, use host=1.1.1.1, or host=1.1.1.0/255.255.255.0. Thanks, --Warren Selby On Jan 12, 2010, at 11:16 AM, Aggio Alberto alberto.ag...@loquendo.com wrote: Hi guys, I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC, with eth0 set to address 192.168.1.1 (NATted over public network, with address 89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) Then I have configured an account as following: [999] type=friend username=999 host=dynamic port=5080 context=sipfrom nat=no canreinvite=no call-limit=8 videosupport=no disallow=all allow=alaw qualify=15000 So far, so good. Now, I have an internal process (onto Linux PC) which is a SIP endpoint and should register to Asterisk as 1.1.1.1:5080, but an external entity (i.e. a SIP endpoint over public Internet) is trying to register to Asterisk as 9...@89.x.y.z:5060 and the registration SUCCEEDS! When I launch the CLI command sip show peers, I see a row like this: 999/9991.1.1.1 5060 OK (3 ms) 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? Thanks for your help! Regards, Alberto Aggio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Minimal Asterisk Web Interface?
I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change my entire OS just for the GUI either (aka AstLinux). Is there anything out there? I'd like to have only a small set of features, primarily the configuration of extensions, routing(in/out), trunks, and ring groups. I welcome your suggestions. :-) Tim -- _ -- 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] VMs IMAP Storage
Hi, is it possible to store a VM in multiple mailboxes ? if not; would it be right to file a RFE so that you could specify on imapuser something like: imapuser=us...@domain.comus...@domain.com like you can with SIP, sounds etc ? Would make it very nice indeed for shared mailboxes. Thoughts ? -- Thanks, Phil -- _ -- 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] SIP Security
- Juan C. Villa juan...@villafam.com wrote: Hey guys, I've been running asterisk on my server for some time now (currently running Asterisk 1.6.2.0). I am having security issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid=blah XX host=dynamic nat=yes canreinvite=no mailbox=1...@default registertrying=yes [testuser] type=friend secret=blah callerid=blah X host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default [testuser2] type=friend username=testuser2 secret= callerid=blah blah host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default Someone is able to connect to my server and make a call since they can access the default context. What should I do? Thanks guys! http://lists.digium.com/mailman/listinfo/asterisk-users http://blogs.digium.com/2009/03/28/sip-security/ -- Thanks, Phil -- _ -- 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] Send 503 or 603 error after answer()
jonas kellens wrote: Is it possible in the Asterisk dialplan to send a 503 Service Unavailable of 603 Decline after having answered the call with Answer() in the dialplan ?? No. Answer generates (for a SIP channel) a '200 OK', which is a final response. You cannot send any further final responses for that INVITE. You need to not answer the call until you really want to answer it and keep it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimal Asterisk Web Interface?
Hi Tim, On Tue, Jan 12, 2010 at 6:56 PM, Tim Nelson tnel...@fudnet.net wrote: I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change my entire OS just for the GUI either (aka AstLinux). Is there anything out there? I'd like to have only a small set of features, primarily the configuration of extensions, routing(in/out), trunks, and ring groups. I welcome your suggestions. :-) I work on a project called AskoziaPBX which may be what you're looking for. It's for embedded devices (min. 64MB RAM, 200MHz) and aimed at beginners so the GUI is quite simple. We also use different terminology (trunks are 'providers' and extensions are 'phones'). We're not yet into named releases but are nearing release candidates for 2.0 (1.0 was based on FreeBSD and Asterisk 1.4, we've now moved to Linux and Asterisk 1.6.1). There's a Live CD you can try on generic x86 hardware, and firmware distributions for products we've specifically tested. Complete firmwares images are less than 15MB. More info here (incomplete site for 2.0): http://2.askozia.com/ Old 1.0 site: http://www.askozia.com/pbx Regards, -Michael -- _ -- 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 Security
Lets just say that you turned off the security ... [general] context=default ; Default context for incoming calls so everyone that can connect to your IP port 5060 UDP can access default context... why would you allow this context to place outgoing calls then ? secret=blah also you think the bots don't know this password ??? Martin On Tue, Jan 12, 2010 at 11:43 AM, Juan C. Villa juan...@villafam.com wrote: Hey guys, I've been running asterisk on my server for some time now (currently running Asterisk 1.6.2.0). I am having security issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid=blah XX host=dynamic nat=yes canreinvite=no mailbox=1...@default registertrying=yes [testuser] type=friend secret=blah callerid=blah X host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default [testuser2] type=friend username=testuser2 secret= callerid=blah blah host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default Someone is able to connect to my server and make a call since they can access the default context. What should I do? Thanks guys! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Security
Martin, I changed all the passwords to blah so I would not reveal them on this email. The password if much more complex than that. It appears that my problem was that I was allowing guest calls. I have beefed up the security, activated fail2ban, along with other things. But thanks anyways! Thanks a ton to Phil who pointed me in the right direction! On Tue, 2010-01-12 at 12:08 -0600, Martin wrote: Lets just say that you turned off the security ... [general] context=default ; Default context for incoming calls so everyone that can connect to your IP port 5060 UDP can access default context... why would you allow this context to place outgoing calls then ? secret=blah also you think the bots don't know this password ??? Martin On Tue, Jan 12, 2010 at 11:43 AM, Juan C. Villa juan...@villafam.com wrote: Hey guys, I've been running asterisk on my server for some time now (currently running Asterisk 1.6.2.0). I am having security issues with my SIP accounts. Unauthorized people have been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid=blah XX host=dynamic nat=yes canreinvite=no mailbox=1...@default registertrying=yes [testuser] type=friend secret=blah callerid=blah X host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default [testuser2] type=friend username=testuser2 secret= callerid=blah blah host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default Someone is able to connect to my server and make a call since they can access the default context. What should I do? Thanks guys! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimal Asterisk Web Interface?
On Tue, Jan 12, 2010 at 08:56:05PM +0300, Tim Nelson wrote: I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change my entire OS just for the GUI either (aka AstLinux). Is there anything out there? I'd like to have only a small set of features, primarily the configuration of extensions, routing(in/out), trunks, and ring groups. I welcome your suggestions. :-) http://svn.asterisk.org/svn/asterisk-gui Designed especially for a minimal, embedded system. Runs no extra daemon besides Asterisk. That said, the no-daemon approach can be a limitation. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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] Send 503 or 603 error after answer()
Thank you for your answer. So if I use early media (not putting answer() at the beginning of my dialplan), how can I send a 503 or 603 from the dialplan ?? Kind regards, Jonas. On Tue, 2010-01-12 at 12:05 -0600, Kevin P. Fleming wrote: jonas kellens wrote: Is it possible in the Asterisk dialplan to send a 503 Service Unavailable of 603 Decline after having answered the call with Answer() in the dialplan ?? No. Answer generates (for a SIP channel) a '200 OK', which is a final response. You cannot send any further final responses for that INVITE. You need to not answer the call until you really want to answer it and keep it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
snip But then the other peer says: -- Called *31#w06123456...@xs4all-out -- SIP/xs4all-out-0234 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION' Anyone an idea where i should look, or how i should change it, so that i do get a wait before sending the rest of the number to the sip peer. /snip I don't have an answer for this but am holding my breath that *someone* does. I ran into a similar situation (dial a number, then wait, then dial an extension via SIP to PSTN) a few weeks ago and never figured out a resolution... My THOUGHT is that you would have to manually inject the DTMF into the stream somehow after the SIP provider connects the call... Thanks Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, January 12, 2010 12:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Inserting a wait in a sip dial snip But then the other peer says: -- Called *31#w06123456...@xs4all-out -- SIP/xs4all-out-0234 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION' Anyone an idea where i should look, or how i should change it, so that i do get a wait before sending the rest of the number to the sip peer. /snip I don't have an answer for this but am holding my breath that *someone* does. I ran into a similar situation (dial a number, then wait, then dial an extension via SIP to PSTN) a few weeks ago and never figured out a resolution... My THOUGHT is that you would have to manually inject the DTMF into the stream somehow after the SIP provider connects the call... Thanks Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Inserting a wait in a sip dial
The problem is only that, it first needs to dial *31#, then wait 1 sec or so, and then dial the number. So it would be needed that its Dial(SIP/*31#w061234123412) But this doesnt seem to work. Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... Regards, Evert -- _ -- 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] Problem logs queue_log-mysql
Hello! I'm trying to registers events of queues in /var/log/asterisk/queue_log and Mysql database .I have configured realtime queue_log on MySQL and works well, but /var/log/asterisk/queue_log file is empty, since you're not registering events of queues. Removing extconfig.conf configurations (queue_log = mysql,general), /var/log/asterisk/queue_log works well, events logs on /var/log/asterisk/queue_log . With extconfig.conf configurations no events logs on /var/log/asterisk/queue_log. What happens?? My asterisk version is 1.6.1.11. addons 1.6.1.2 res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = userX dbpass = passX dbport = 3306 dbsock = /tmp/mysql.sock -- extconfig.conf [settings] queue_log = mysql,general logger.conf [general] queue_log = yes queue_log_name = queue_log Thanks, Best regards!! Cristian Arguello. __ Información de ESET NOD32 Antivirus, versión de la base de firmas de virus 4765 (20100112) __ ESET NOD32 Antivirus ha comprobado este mensaje. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
This doesn't work? Dial(SIP/*31#ww061234123412) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ev...@disruptor.nl Sent: Tuesday, January 12, 2010 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inserting a wait in a sip dial The problem is only that, it first needs to dial *31#, then wait 1 sec or so, and then dial the number. So it would be needed that its Dial(SIP/*31#w061234123412) But this doesnt seem to work. Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... Regards, Evert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
Ok my problem is solved now, it was easyer fixed by adding: Set(CALLERPRES()=unavailable) That did exactly the same as the *31# would have done. So for me the problem is solved. The problem is only that, it first needs to dial *31#, then wait 1 sec or so, and then dial the number. So it would be needed that its Dial(SIP/*31#w061234123412) But this doesnt seem to work. Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a 1/2 second delay before dialing, ww1234 a 1 second delay, etc. Try it with 2 or 3 w's instead of 1... Regards, Evert Regards, Evert -- _ -- 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
Jeff LaCoursiere wrote: I thought the discussion was about how an IT guy with no previous asterisk experience could get up and running the fastest. By FAR that answer is to No, actually he said, This is currently still at a proof of concept stage. By whose estimation? To even get that far with asterisk requires a lot of reading and experience. It took me several weeks to get my first Me too, and I enjoyed immensely! But then again, I'm a geek. Doug -- _ -- 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] Minimal Asterisk Web Interface?
- Michael Iedema mich...@askozia.com wrote: Hi Tim, On Tue, Jan 12, 2010 at 6:56 PM, Tim Nelson tnel...@fudnet.net wrote: I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change my entire OS just for the GUI either (aka AstLinux). Is there anything out there? I'd like to have only a small set of features, primarily the configuration of extensions, routing(in/out), trunks, and ring groups. I welcome your suggestions. :-) I work on a project called AskoziaPBX which may be what you're looking for. It's for embedded devices (min. 64MB RAM, 200MHz) and aimed at beginners so the GUI is quite simple. We also use different terminology (trunks are 'providers' and extensions are 'phones'). We're not yet into named releases but are nearing release candidates for 2.0 (1.0 was based on FreeBSD and Asterisk 1.4, we've now moved to Linux and Asterisk 1.6.1). There's a Live CD you can try on generic x86 hardware, and firmware distributions for products we've specifically tested. Complete firmwares images are less than 15MB. More info here (incomplete site for 2.0): http://2.askozia.com/ Old 1.0 site: http://www.askozia.com/pbx Regards, -Michael The problem is, I don't want to replace my underlying OS. I just need the web interface. That being said, I have used Askozia in the past and may use it in a future project. Thank you for the suggestion! Tim -- _ -- 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] Minimal Asterisk Web Interface?
- Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Jan 12, 2010 at 08:56:05PM +0300, Tim Nelson wrote: I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change my entire OS just for the GUI either (aka AstLinux). Is there anything out there? I'd like to have only a small set of features, primarily the configuration of extensions, routing(in/out), trunks, and ring groups. I welcome your suggestions. :-) http://svn.asterisk.org/svn/asterisk-gui Designed especially for a minimal, embedded system. Runs no extra daemon besides Asterisk. That said, the no-daemon approach can be a limitation. Yesss. That is exactly what I needed! \o/ Installed, it uses around 1.7MB of disk space, requires no external dependencies (other than Asterisk), and appears to more than cover my needs. Thank you Tzafrir Of course, if there are any additional minimal GUIs available, I'd be more than interested to have a look at those also. One can never have too many options... :-) Tim -- _ -- 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] Inserting a wait in a sip dial
snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as well (assuming the provider honors it)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
David Gibbons wrote: snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as well (assuming the provider honors it)? 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). In general, dial-string feature codes like this are not used on 'intelligent signaling' channels like SIP and ISDN; there are nearly always other, proper, ways to get the desired effect. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
snip 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). /snip Thanks Kevin, that's what I figured (though not quite so concisely)... Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? So: 1. dial 12345 2. connect SIP provider to * extension 3. wait 2 seconds programmatically 3. inject 567 DTMF tones into channel to signal remote PBX of extension to dial I'm hoping there's another way to skin this cat. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
2010/1/12 Robert Lister r...@lentil.org: 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.) Up until a year ago when we got our new useless Alcatel system all our lines were analogue. When we got the Alcatel we got told Everything has gone digital so we now have a ISDN PRI. I'm now seeing this is actually far from the truth but never mind Also you will need to think if you want to support analogue devices such as modems/fax machines etc. If the system can take faxes then fine, We already use Hylafax on a separate analogue line currently, which is were it will remain, unless I can find a good reason to change. If Asteriks can identify a fax coming in on the main line and do the right thing then that would be a neat feature, but its in the end of the world if its not there. 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) Starting from scratch, I'm not sure I trust soft phones enough, But it would be cheap and the project has very little budget currently! What sort of support do you require for the system / handsets? Do you need CTI integration / soft phones / headsets etc? Yes this is vital its the one of the big things we miss since we got the new Alcatel (The Alcatel crashes every 2 days if we switch the CTI on!) Headset vital. How many lines in total are coming in to the system? Currently 1 ISDN PRI I think but I can't see anything Asterks should not be able to handle. we used to have 48 Analogue lines but I've not seen the office having more than 5 calls at the same time in years. Do you need hotdesk users or are they all based at the same desks every day? Totally HotDesk 24x7 phones are always in use. We don't currently have personal extensions but this would be a nice feature What are the requirements for redundancy/failover? (ranging from 'none' to 'magic failover between two sites') Fallover would be nice again we don't have any currently. we would also like people to be able to log in and take calls from home from time to time when we get really busy I'm looking at AsterksNow/TrixBox but I'm a ubuntu guy (whole office is running on Ubuntu for our desktops) so if the phones run that too it would mean everything was the same, but if the simplest solution is different then fine. I do need a GUI that is easy to deal with, ie adding users, groups, queues etc. 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.) I've got a basic idea what I need, I'm just trying to work out a demo to get the idea of the board past management (Without causing too much trouble) 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. It shows how good our old STS system really was! 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. Peter. 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 -- _ -- 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] Multi-Tenant Parking
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD Sent: Tuesday, 12 January 2010 17:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-Tenant Parking Should that not say parkinglot and not parkinglog in features.conf? It should – but that’s not a cut and paste, as the asterisk setup is on a separate, non-connected network, and I just retyped it out – not cut/paste. It’s spelt correctly in the real system (typo on here!) IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- 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] Multi-Tenant Parking
Have you looked at this? http://www.google.com/#q=app_valetparking I have - but would rather use the inbuilt functionality if possible before resorting to third-party code... IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- 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] Inserting a wait in a sip dial SOLVED (kind of)
snip Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? /snip Well, due to the lack of responses, either I missed something obvious or nobody cares. I'm really hoping I didn't miss something obvious... :). In any event, I got curious of my own old question and hacked out a work around: 0. Assume your extension is dumped into context 'mycontext' 1. You dial an internal extension 2. * Dials an external number (presumably another PBX device) 3. When the remote device answers, both parties are dumped into the DTMFworkaround context 4. The called party has its DTMF mode set to inband so that the tones are played out loud 4.5. Meanwhile, the calling party is dumped into an empty meeting conference that is used soley to bridge these two legs 5. When the tones are done, the called party is dumped into the bridged conference. 6. When the caller hangs up, the conference boots the callee code [dtmfworkaround] exten = 6534,1,Goto(dtmfworkaround|6536|1) exten = 6534,2,Goto(dtmfworkaround|6535|1) exten = 6535,1,Answer() exten = 6535,n,Wait(1) exten = 6535,n,SIPDTMFMode(inband) exten = 6535,n,SendDTMF(1234) exten = 6535,n,MeetMe(101|MFqx|1234) exten = 6536,1,Answer() exten = 6536,n,MeetMe(101|MFqxA|1234) [mycontext] exten = 658,1,Dial(SIP/486,15,rG(dtmfworkaround^6534^1)) /code -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
RFC2833 is carried in RTP like the audio stream. However, it uses a
different payload type from the RTP packets used to transport the
audio. If you did an RTP capture you would be able to see the RFC2833
events (which correspond to DTMF presses).
The SIP debug, however, will tell you if the remote end is configured
to use RFC2833 or not. That's why I was telling you to look for
telephone-event in the INVITE from your provider. Keep in mind SIP
(most likely) runs over UDP between you and your provider, not TCP.
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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[asterisk-users] AudioCodes MP-114 - SAS (Stand Alone Survivability) configuration
I have AudioCodes MP-114 and I'm trying to configure SAS (Stand Alone Survivability); when Asterisk is down the MediaPack gateway should forward the call IN/OUT through the gateway (without asterisk in the middle), but it is not working. I'm working with tech. support from the source I purchase the unit from they we are just emailing back and forth and the unit is still not working. Can anybody share settings how to configure Stand Alone Survivability mode in AudioCodes MP-1xx ? -- Joseph -- _ -- 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] Xorcom 32 channel FXS gateway
Anyone on the list ever used it? I'm trying to quote a system with 192 analog ports, one of the options are the Xorcom 32 channel FXS USB Channel Banks. Any input would be appreciated. TIA -- _ -- 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] Send 503 or 603 error after answer()
jonas kellens wrote: So if I use early media (not putting answer() at the beginning of my dialplan), how can I send a 503 or 603 from the dialplan ?? By using the proper method of canceling the call... Busy, Congestion, or an explicit cause code passed to Hangup. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Xorcom 32 channel FXS gateway
On Tue, 2010-01-12 at 18:05 -0500, C F wrote: Anyone on the list ever used it? I'm trying to quote a system with 192 analog ports, one of the options are the Xorcom 32 channel FXS USB Channel Banks. Any input would be appreciated. I have used Astribanks for a while now and they are usually very stable. The only thing to worry about is that if you change the order the units are connected to the USB ports then you will have a mess on your hands. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Xorcom 32 channel FXS gateway
You can address the order of detection problem using udev rules... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Tuesday, January 12, 2010 6:53 PM To: Asterisk Users List Subject: Re: [asterisk-users] Xorcom 32 channel FXS gateway On Tue, 2010-01-12 at 18:05 -0500, C F wrote: Anyone on the list ever used it? I'm trying to quote a system with 192 analog ports, one of the options are the Xorcom 32 channel FXS USB Channel Banks. Any input would be appreciated. I have used Astribanks for a while now and they are usually very stable. The only thing to worry about is that if you change the order the units are connected to the USB ports then you will have a mess on your hands. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] Inserting a wait in a sip dial SOLVED (kind of)
Dave-- I remember adding a feature a long time ago for snoms, to the source code, to send dtmf out for some button press on a snom phone, in the 'outward' direction, I think to activate a feature or somesuch. (Boy, is my memory hazy!) At any rate, I was able to inject dtmf, but I had to do it in the source. AFAICT, there is no app that do this explicitly; and Murphy's Law would state that even if a dialplan app existed, it would not get run at the time you need to be run. So, if you found a workaround, and it works, it won't matter how pretty it is. Magic is Magic. And speaking of Murphy's Law: Enjoy it while it lasts, because, sure as death and taxes, someone will fix a bug somewhere, and you'll lose an undocumented feature ;) murf On Tue, Jan 12, 2010 at 2:31 PM, David Gibbons d...@videon-central.comwrote: snip Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? /snip Well, due to the lack of responses, either I missed something obvious or nobody cares. I'm really hoping I didn't miss something obvious... :). In any event, I got curious of my own old question and hacked out a work around: 0. Assume your extension is dumped into context 'mycontext' 1. You dial an internal extension 2. * Dials an external number (presumably another PBX device) 3. When the remote device answers, both parties are dumped into the DTMFworkaround context 4. The called party has its DTMF mode set to inband so that the tones are played out loud 4.5. Meanwhile, the calling party is dumped into an empty meeting conference that is used soley to bridge these two legs 5. When the tones are done, the called party is dumped into the bridged conference. 6. When the caller hangs up, the conference boots the callee code [dtmfworkaround] exten = 6534,1,Goto(dtmfworkaround|6536|1) exten = 6534,2,Goto(dtmfworkaround|6535|1) exten = 6535,1,Answer() exten = 6535,n,Wait(1) exten = 6535,n,SIPDTMFMode(inband) exten = 6535,n,SendDTMF(1234) exten = 6535,n,MeetMe(101|MFqx|1234) exten = 6536,1,Answer() exten = 6536,n,MeetMe(101|MFqxA|1234) [mycontext] exten = 658,1,Dial(SIP/486,15,rG(dtmfworkaround^6534^1)) /code -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy ParseTree Corp -- _ -- 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] Inserting a wait in a sip dial SOLVED (kind of)
If you need to inject dtmf tones or sound into an existing channel you can use chanspy with option w. I play sound files using the AMI to originate a call to an extension that does chanspy on one leg and a playback on the other. I use channel variables to say which channel to play to and which sound file to play. SendDTMF or Playtones should be able to inject tones. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 12, 2010, at 4:19 PM, Steve Murphy wrote: Dave-- I remember adding a feature a long time ago for snoms, to the source code, to send dtmf out for some button press on a snom phone, in the 'outward' direction, I think to activate a feature or somesuch. (Boy, is my memory hazy!) At any rate, I was able to inject dtmf, but I had to do it in the source. AFAICT, there is no app that do this explicitly; and Murphy's Law would state that even if a dialplan app existed, it would not get run at the time you need to be run. So, if you found a workaround, and it works, it won't matter how pretty it is. Magic is Magic. And speaking of Murphy's Law: Enjoy it while it lasts, because, sure as death and taxes, someone will fix a bug somewhere, and you'll lose an undocumented feature ;) murf On Tue, Jan 12, 2010 at 2:31 PM, David Gibbons d...@videon-central.com wrote: snip Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? /snip Well, due to the lack of responses, either I missed something obvious or nobody cares. I'm really hoping I didn't miss something obvious... :). In any event, I got curious of my own old question and hacked out a work around: 0. Assume your extension is dumped into context 'mycontext' 1. You dial an internal extension 2. * Dials an external number (presumably another PBX device) 3. When the remote device answers, both parties are dumped into the DTMFworkaround context 4. The called party has its DTMF mode set to inband so that the tones are played out loud 4.5. Meanwhile, the calling party is dumped into an empty meeting conference that is used soley to bridge these two legs 5. When the tones are done, the called party is dumped into the bridged conference. 6. When the caller hangs up, the conference boots the callee code [dtmfworkaround] exten = 6534,1,Goto(dtmfworkaround|6536|1) exten = 6534,2,Goto(dtmfworkaround|6535|1) exten = 6535,1,Answer() exten = 6535,n,Wait(1) exten = 6535,n,SIPDTMFMode(inband) exten = 6535,n,SendDTMF(1234) exten = 6535,n,MeetMe(101|MFqx|1234) exten = 6536,1,Answer() exten = 6536,n,MeetMe(101|MFqxA|1234) [mycontext] exten = 658,1,Dial(SIP/486,15,rG(dtmfworkaround^6534^1)) /code -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy ParseTree Corp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Xorcom 32 channel FXS gateway
Any echo issues using FXS ports? On Tue, Jan 12, 2010 at 6:53 PM, Carlos Chavez cur...@telecomabmex.com wrote: On Tue, 2010-01-12 at 18:05 -0500, C F wrote: Anyone on the list ever used it? I'm trying to quote a system with 192 analog ports, one of the options are the Xorcom 32 channel FXS USB Channel Banks. Any input would be appreciated. I have used Astribanks for a while now and they are usually very stable. The only thing to worry about is that if you change the orderhttp://mail.google.com/mail/images/cleardot.gif http://mail.google.com/mail/images/cleardot.gif the units are connected to the USB ports then you will have a mess on your hands. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Parking (HALF SOLVED)
I have found that this seems to be a functional difference between the Park() and the ParkAndAnnounce() functions. Park() respects the parking lot specification, yet ParkAndAnnounce() does not respect the fact that you’ve tried to arbitrarily set the parking lot. The code below “works” as designed when the Park() function is used instead. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD Sent: Tuesday, 12 January 2010 17:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-Tenant Parking Should that not say parkinglot and not parkinglog in features.conf? It should – but that’s not a cut and paste, as the asterisk setup is on a separate, non-connected network, and I just retyped it out – not cut/paste. It’s spelt correctly in the real system (typo on here!) IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- 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?
2010/1/12 Lenz Emilitri lenz.lo...@gmail.com: You can list phones directly as static members of the queue. i know i can configure the queue.conf and agents.conf to add queue name and queue members by hand. Could i use functions to create queue name and add queue members dynamiclly. because i want to create a call center use asterisk, which users can register their own call number on the web site. also they can add several service phone numbers along with a fix extension (like:1), the phone numbers are customer service number, when it's customer dial the call number and press extension 1, one member should answer the caller. so, when configured on the web. like: extension 1:12345, 12346,12347,12348,12349 when finished the data above should stored in the database, when user call in and press 1. i should create a queue and add 12345, 12346,12347,12348,12349 to the queue. is this possible? this is generally sub.optimal because if. e.g. an agent of yours is home sick, her phone will be ringing and you'll be wasting caller time. Also by tracking logins and logoffs you can measure agent productivity, and this is pretty useful in most environments. because all the phones are office phone, if one phone can't response as on people there , with 15 secs timeout, it will select another phone in the queue according the strategy. so it doesn't matter. l. 2010/1/12 Zhang Shukun bit...@gmail.com Thank you! it's very helpful. now i have another question: in asterisk, each agent should login first and then can response to the caller. but i don't want to the login action. i need agent shold response directly without login first. how should i do ? can users in sip.conf to be agents? so it can login persistently on a phone. -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- 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?
2010/1/13 Robert Lister r...@lentil.org: 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 Thank you! you explained very clear above about the two concept! 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. is there some function used to login a agent automaticlly like agentlogin(agentname,agentpassword,phonenumber)? 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 -- Best regards, Sucan -- _ -- 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 Voicemail Issue
I have several extensions in the Central Timezone, the Server is in the Eastern Timezone. all the voicemail files have a datetimestamp of EST not of the tz= option under the usermail ... voicemail.conf under [general] tz=EST under [default] mailbox_a,password,,,tz=CST6CDT mailbox_b,password,,,tz=CST6CDT mailbox_c,password,,,tz=EST Server Time is EST Now if a the owner of mailbox_b calls mailbox_a extension, and leaves a message, the message has a timestamp of EST not CST ?? -- _ -- 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 Voicemail Issue
ok, I figured it out.. tz=zonename from zonemessages all fixed. - Original Message - From: William Stillwell ( Lists ) To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, January 12, 2010 9:59 PM Subject: [asterisk-users] Odd Voicemail Issue I have several extensions in the Central Timezone, the Server is in the Eastern Timezone. all the voicemail files have a datetimestamp of EST not of the tz= option under the usermail . voicemail.conf under [general] tz=EST under [default] mailbox_a,password,,,tz=CST6CDT mailbox_b,password,,,tz=CST6CDT mailbox_c,password,,,tz=EST Server Time is EST Now if a the owner of mailbox_b calls mailbox_a extension, and leaves a message, the message has a timestamp of EST not CST ?? -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Mute Problem
Hey Yall I have an interesting situation. When a person is on the phone (Polycom 501) and another call hits the phone the phone mutes on the users side not the person outside the asterisk system. It stays mute until the call goes to voicemail. Its like the beep we should get from the call waiting has turned into dead silence. Any ideas Thanks Michael D Mosier __ Information from ESET Smart Security, version of virus signature database 4628 (20091122) __ The message was checked by ESET Smart Security. http://www.eset.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID on Indian PSTN is not working.
Hi, I got a solution for this problem from Freepbx forumhttp://www.freepbx.org/forum/freepbx/users/caller-id-not-working#comment-23520. Is anybody know about this DTMF to FSK converter? Is this solution solve my problem? Any way I will try it and get back. -- Thanks, Arun S System Administrator. Cabot Solutions www.cabotsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
Dear All I have Asterisk 1.4 installed on my Debian server . I am considering upgrading my Asterisk to the latest version (1.6) . Can you please let me know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk 1.6 ? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
At 23:52 1/10/2010, Doug wrote: At 15:33 1/7/2010, Tzafrir Cohen wrote: On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote: At 00:22 1/7/2010, Tzafrir Cohen wrote: On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote: At 16:49 1/5/2010, Tzafrir Cohen wrote: On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote: Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. What version of SpanDSP do you use? spandsp-0.0.6pre12.tgz and: libtiff-3.8.2-7.el5_3.4 libtiff-devel-3.8.2-7.el5_3.4 Which do you recommend? What errors do you get? I'm using a backport of app_fax.c and it works well. Do you have the link for the C source? app_fax.c from: https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons/trun k/app-spandsp/ Just remove the '#include ../addon_version.h line, and the single include used from it (AGX_AST_ADDON_VERSION). Could you please elaborate on the above? If I comment out: /* Commented out #include ../addon_version.h */ Are you saying that I need to comment out this? /* ast_log(LOG_NOTICE, app_fax %s using spandsp %s\n, AGX_AST_ADDON_VERSION, SPANDSP_RELEASE_DATETIME_STRING ); */ When I compile, I get a bunch of errors: # make Scanning dependencies of target app_fax [ 50%] Building C object CMakeFiles/app_fax.dir/app_fax.o /usr/src/asterisk/app_fax/app_fax.c: In function âphase_e_handlerâ: /usr/src/asterisk/app_fax/app_fax.c:202: error: missing terminating character /usr/src/asterisk/app_fax/app_fax.c:203: error: expected expression before â%â token /usr/src/asterisk/app_fax/app_fax.c:203: error: stray â\â in program /usr/src/asterisk/app_fax/app_fax.c:203: error: stray â\â in program /usr/src/asterisk/app_fax/app_fax.c:203: error: stray â\â in program /usr/src/asterisk/app_fax/app_fax.c:203: error: stray â\â in program /usr/src/asterisk/app_fax/app_fax.c:203: error: stray â\â in program /usr/src/asterisk/app_fax/app_fax.c:203: error: missing terminating character /usr/src/asterisk/app_fax/app_fax.c:688:1: error: unterminated argument list invoking macro fax_log /usr/src/asterisk/app_fax/app_fax.c: In function âphase_d_handlerâ: /usr/src/asterisk/app_fax/app_fax.c:240: error: âfax_logâ undeclared (first use in this function) /usr/src/asterisk/app_fax/app_fax.c:240: error: (Each undeclared identifier is reported only once /usr/src/asterisk/app_fax/app_fax.c:240: error: for each function it appears in.) /usr/src/asterisk/app_fax/app_fax.c:240: error: expected â;â at end of input /usr/src/asterisk/app_fax/app_fax.c:240: error: expected declaration or statement at end of input /usr/src/asterisk/app_fax/app_fax.c:240: error: expected declaration or statement at end of input make[2]: *** [CMakeFiles/app_fax.dir/app_fax.o] Error 1 make[1]: *** [CMakeFiles/app_fax.dir/all] Error 2 make: *** [all] Error 2 Any ideas? Well, you might consider using nano with the -w option so when you copy and paste into nano, your source code won't wordwrap. -- _ -- 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] Inserting a wait in a sip dial
Kevin P. Fleming wrote: David Gibbons wrote: snip This doesn't work? Dial(SIP/*31#ww061234123412) /snip When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as well (assuming the provider honors it)? 'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). hmm, I use 'w' on ISDN channels (libpri) to signal sending complete, like Dial(DAHDI/g1/0123456w). But I did not know that 'w' means actually 'wait'. Regards Hans -- _ -- 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] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Tue, 2010-01-12 at 16:52 -0500, Kristian Kielhofner wrote:
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
RFC2833 is carried in RTP like the audio stream. However, it uses a
different payload type from the RTP packets used to transport the
audio. If you did an RTP capture you would be able to see the RFC2833
events (which correspond to DTMF presses).
Thanks for that. Looking at the RTP packets I can see two types as you
point out. The first appears to be the audio:
Real-Time Transport Protocol
10.. = Version: RFC 1889 Version (2)
Payload type: ITU-T G.711 PCMU (0)
And as you say, the DTMF events are clear to see:
RFC 2833 RTP Event
Event ID: DTMF One 1 (1)
..00 1010 = Volume: 10
So, as these can be seen in the stream, do I need to tell Asterisk to
detect these? Does it not do that when I set: dtmfmode=rfc2833
???
The SIP debug, however, will tell you if the remote end is configured
to use RFC2833 or not. That's why I was telling you to look for
telephone-event in the INVITE from your provider. Keep in mind SIP
(most likely) runs over UDP between you and your provider, not TCP.
I see a 'telephone-event' :
a=rtpmap:101 telephone-event/8000
buried in the chunk below. but I have to be honest, SIP is new to me so
I'm not sure of myself with this:
v=0
o=root 27089 27089 IN IP4 217.10.69.13
s=session
c=IN IP4 217.10.69.13
t=0 0
m=audio 19990 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
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