[asterisk-users] Asterisk V1.6.0 SVN debug WARNING(6830) a bug or deliberate?
Hi lists,
Does anyone know if the following error message (from a debug screen) was
a
deliberate change from the behavior in asterisk V1.4.18 or just an
overlooked
parsing error in progressing to V1.6.0? Since, in this case, the string (Hi
there)
is quoted, it doesn't seem as though the parser should take notice about
about the
interior of a 'word'. However, if it is deliberate, then so be it. (a yellow
NOTICE
would be more soothing than a red WARNING) :-)
Gerald Harshany
WARNING(6830): pbx.c:7557 pbx_builtin_setvar: Please avoid unnecessary
spaces on variables as it may lead to unexpected results
('DB(Knowselgreat/Hi there)' set to ' myfile ').
Using current Asterisk version: SVN-branch-1.6.0-r114304 (on Ubuntu) and
Zaptel current SVN 1.4
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[asterisk-users] AddQueueMember() and PersistentMembers
Hi,
I'm trying to use AddQueueMember() to add a member to a queue and trying to
make this logged member in the queue between reloads and restarts of
asterisk.
I configure en queues.conf:
[general]
Persistentmembers=yes
And Extensions.conf:
exten=
*01,1,AddQueueMember(queue_name,Local/${CALLERID(num)[EMAIL PROTECTED],penalty);
When I log with AddQueueMember to any queue and stop and load asterisk
again, the database entry disappear. Is this a normal behavior?
I tried to look at the code in app_queue.c and check at
reload_queue_member() function, that function does not found the database
entry /Queues/PersistentMembers/queue_name.
Am I wrong? Any help?
Thanks.
Alejandro Guercio
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Re: [asterisk-users] Newbie Queue: greetings when first joiningqueue
Check the number of calls waiting in the queue, then play the message
if
more than 0
example code (written in the TBird IDE)
Exten = 100,1,Answer()
Exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
Exten = 100,n,GotoIf($[${NumWaiting} = 0]?JoinQueue)
Exten = 100,n,PlayBack(MyMessage)
Exten = 100,n(JoinQueue),Queue(MyQueueName)
Exten = 100,n,Hangup()
So, if there are no members in the queue, jump directly to the queue
application, otherwise play the message first.
Thanks Julian and it certainly works.
I have got another question if I may.
If there is just one agent in the queue and he put on Do-Not-Disturb,
certainly in this case the queue count will be zero but I would still
like Asterisk to play back Welcome to XYZ, your call is important to us
... please stay on the line, the logic above would fail to play back
the intro message.
I thought about trapping DIALSTATUS but if there is actually no dial
cmd, how can I trap the DND then and play back the message again?
Any thoughts?
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[asterisk-users] zap not coming online on fedora 8
Hi All; So what should I do now to remove that bug? Any advise? Regards Bilal --- Hence the reason I kept CCing your off-list emails back to the list. Guys like Tzafrir are aces. (Just a reminder that if this is indeed the case then this is a bug inflicted by me and fixed later by sruffell. Makes me feel like the glassmaker sending the kid to break the window) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Polycom 3.0
How do they get away with that? On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html -Jon - Original Message - From: Darrick Hartman (lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 28, 2008 5:18:26 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 Andreas van dem Helge wrote: Anyone have a download link for 3.0 SIP firmware? If you are going to say ask polycom or ask your vendor don't even waste your time posting. I've asked the Nazis and they'll probably take 1 week. Suggest you get a different vendor then. I got a response from mine within a few hours. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided 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] Hyperthreading and multicore
Asterisk didn't benefit much from having HT enabled on a P4 with HT capability. There are several things that make a difference when optimizing for a specific processor in order to take advantage of its features. Gcc version used to build asterisk (and the system in general) and compile flags can make a big difference A lot of the ready made solutions use very generic optimization as they are trying to be compatible with a wide range of cpu core's and architectures. This has the advantage of having a single binary image to distribute but you pay for it in terms of performance. In most cases the performance penalty is not noticable in small/home installations but you start to notice it when you push the system to its resource limit (i.e cpu, memory,pci bus access etc) either because you handle a lot of calls or your system is resourse limited i.e embedded boards. So in general if you need to get the maximum performance out of a system, make sure you build asterisk tuned for that system and not a generic build. Running code with 486 instruction set, with command scheduling for pentium its not going to give you max performance regardless of the fact that your cpu/core supports HT or not. Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.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] PRI CallerID - leading zero added
Hello List!
We have problems setting the right caller id on outgoing calls. The
Asterisk Pbx is located
in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the
local telefon number
40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID
numbers available.
The telco is aspecting a 3 digit long Callerid from us, for example
like 710, for the extension 10.
Therefore the caller id is set up like 7+double digit extension number.
We set the caller id in asterisk to the 3 digits as they told us, but
somehow a leading zero is added to
the callerid on my side. Callerid received by the Telco is then 0710
and the Callerid on outgoing calls
is ignored by the Telco and set to default 4078700.
Below a listing of my settings and dialplan of my asterisk pbx. The E1 line is
correctly set up - the Telco told us, so framing, signaltype and so on
should be right.
The attachment is the output of pri debug span 1 from Asterisk-CLI,
making a call from extension
66 to an Austrian number.
- We cannot determine where the leading zero is added to the caller ID.
- Is there another way to debug this? Not from asterisk-CLI.
Our settup is the following,
Asterisk 1.2.19, zaptel 1.2.18
Digium TE120P T1/E1 card
- zaptel.conf:
loadzone=nl
defaultzone=nl
span=1,1,0,ccs,hdb3,crc4,yellow
bchan=1-15,17-31
dchan=16
- zapata.conf
[trunkgroups]
[channels]
language=en
context=isdn-incoming-e1
switchtype=euroisdn
;internationalprefix =
;nationalprefix =
;localprefix =
overlapdial=yes
signalling=pri_cpe
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
channel = 1-15,17-31
- asterisk extensions.conf macro for dialout over TE120P:
[macro-dialout-isdn]
;arg1=calleridnum (in our case doubledigit extension number), arg2=number dialed
exten = s,1,Set(CALLERID(num)=7${ARG1})
exten = s,2,Answer
exten = s,3,Dial(ZAP/g1/${ARG2})
exten = s,4,Goto(s-${DIALSTATUS}|1)
exten = s-NOANSWER,1,Noop( !!! Isdngruppe g1 = NOANSWER)
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,Noop( !!! Isdngruppe g1 = BUSY)
exten = s-BUSY,2,Busy
exten = s-CHANUNAVAIL,1,Noop( !!! Isdngruppe g1 = CHANUNAVAIL)
exten = s-CHANUNAVAIL,2,Hangup
exten = s-CHANCEL,1,Noop( !!! Isdngruppe g1 = CHANCEL)
exten = s-CHANCEL,2,Hangup
exten = s-CONGESTION,1,Noop( !!! Isdngruppe g1 = CONGESTION)
exten = s-CONGESTION,2,Hangup
exten = s-ANSWER,1,Noop( !!! Isdngruppe g1 = ANSWER)
exten = s-ANSWER,2,Hangup
exten = s-HANGUP,1,Noop( !!! Isdngruppe g1 = HANGUP)
exten = s-HANGUP,2,Hangup
When i make a call from extension 10 the macro is called with that:
exten = _0.,1,Macro(dialout-isdn,10,${EXTEN:1})
Christian Gansberger
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Re: [asterisk-users] realtime queue callers
On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote: How can I get a list of the callers within a specific queue at any given moment? I need to get the caller IDs of all active calls in a queue then send them out via a udp socket to a listening application on the network (the only data I need to send are two fields: current timestamp and caller id of active queue calls). I have almost all the elements to do this except the best method to retrieve all active caller ids from a given queue. I was wondering if someone already did this. I tried writing a script on the server which connects to the Manager API and receives queue events. I'm basically using the AgentCalled event but it seems clumsy to efficiently detect when the call has ended (connect or abandon) and thus update the remote UDP listening app. I also tried another way by guessing which calls are active via tailing and grepping /var/log/asterisk/queue_log. Finally, a third script method tried parsing the output of show queue right after Callers:. Maybe this is all I really need for my purposes (although less efficient and less real-time than the queue events method because I would need to periodically poll the whole queue statistics) but I only get the originating channel and the wait time. I would require correlating the data to the caller's ID. Has anyone already done something similar? A simple example/script/suggestion would be greatly appreciated. I'm not sure that this is what exactly You need, but I have a patch for app_queue that will store and update queue callers (as well as update lots of fields for queue members) in realtime mysql table. This allows to do many requests for current queue state simultenously, and moves load from asterisk to mysql (which can be on separate machine). So, generally to get active callers with all their callerid/channel info You will have to do just SELECT * FROM queue_callers. It's not very finalized, so I haven't yet posted that to Digium for inclusion in next asterisk versions, but I intend to do that in future. It's been working stable on our production for several months. If You're interested, please reply, and I'll try to separate that patch out from other our patches. Currently I have it updated for 1.4.19, but also have some version for 1.4.14 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue callers
--- Atis Lezdins [EMAIL PROTECTED] wrote: On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote: How can I get a list of the callers within a specific queue at any given moment? I need to get the caller IDs of all active calls in a queue then send them out via a udp socket to a listening application on the network (the only data I need to send are two fields: current timestamp and caller id of active queue calls). I have almost all the elements to do this except the best method to retrieve all active caller ids from a given queue. I was wondering if someone already did this. I tried writing a script on the server which connects to the Manager API and receives queue events. I'm basically using the AgentCalled event but it seems clumsy to efficiently detect when the call has ended (connect or abandon) and thus update the remote UDP listening app. I also tried another way by guessing which calls are active via tailing and grepping /var/log/asterisk/queue_log. Finally, a third script method tried parsing the output of show queue right after Callers:. Maybe this is all I really need for my purposes (although less efficient and less real-time than the queue events method because I would need to periodically poll the whole queue statistics) but I only get the originating channel and the wait time. I would require correlating the data to the caller's ID. Has anyone already done something similar? A simple example/script/suggestion would be greatly appreciated. I'm not sure that this is what exactly You need, but I have a patch for app_queue that will store and update queue callers (as well as update lots of fields for queue members) in realtime mysql table. This allows to do many requests for current queue state simultenously, and moves load from asterisk to mysql (which can be on separate machine). So, generally to get active callers with all their callerid/channel info You will have to do just SELECT * FROM queue_callers. It's not very finalized, so I haven't yet posted that to Digium for inclusion in next asterisk versions, but I intend to do that in future. It's been working stable on our production for several months. If You're interested, please reply, and I'll try to separate that patch out from other our patches. Currently I have it updated for 1.4.19, but also have some version for 1.4.14 Thanks Atis. That patch sounds really neat. Hope it gets into * soon. Just a doubt: suppose the mysql daemon dies for some reason. Will the patched app_queue still handle calls and not hang? Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue callers
On Tue, Apr 29, 2008 at 1:22 PM, Vieri [EMAIL PROTECTED] wrote: --- Atis Lezdins [EMAIL PROTECTED] wrote: On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote: How can I get a list of the callers within a specific queue at any given moment? I need to get the caller IDs of all active calls in a queue then send them out via a udp socket to a listening application on the network (the only data I need to send are two fields: current timestamp and caller id of active queue calls). I have almost all the elements to do this except the best method to retrieve all active caller ids from a given queue. I was wondering if someone already did this. I tried writing a script on the server which connects to the Manager API and receives queue events. I'm basically using the AgentCalled event but it seems clumsy to efficiently detect when the call has ended (connect or abandon) and thus update the remote UDP listening app. I also tried another way by guessing which calls are active via tailing and grepping /var/log/asterisk/queue_log. Finally, a third script method tried parsing the output of show queue right after Callers:. Maybe this is all I really need for my purposes (although less efficient and less real-time than the queue events method because I would need to periodically poll the whole queue statistics) but I only get the originating channel and the wait time. I would require correlating the data to the caller's ID. Has anyone already done something similar? A simple example/script/suggestion would be greatly appreciated. I'm not sure that this is what exactly You need, but I have a patch for app_queue that will store and update queue callers (as well as update lots of fields for queue members) in realtime mysql table. This allows to do many requests for current queue state simultenously, and moves load from asterisk to mysql (which can be on separate machine). So, generally to get active callers with all their callerid/channel info You will have to do just SELECT * FROM queue_callers. It's not very finalized, so I haven't yet posted that to Digium for inclusion in next asterisk versions, but I intend to do that in future. It's been working stable on our production for several months. If You're interested, please reply, and I'll try to separate that patch out from other our patches. Currently I have it updated for 1.4.19, but also have some version for 1.4.14 Thanks Atis. That patch sounds really neat. Hope it gets into * soon. Just a doubt: suppose the mysql daemon dies for some reason. Will the patched app_queue still handle calls and not hang? It should, as asterisk throws INSERTs, UPDATEs and DELETEs for changing data (queue callers and queue member status), plus it loads existing queue members trough SELECT (as it's now with realtime queue members, just some extra fields). So, I suppose if MySQL dies in middle of operation, SELECT should fail and Asterisk should just continue with what it has in memory. Btw, You should be able to also use static or dynamic queue members (not realtime) in combination with realtime queue calls. Btw, I never experienced that MySQL dies, it's more often that Asterisk dies. So, are You interested in applying this patch yourself? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided 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] Annoying Sipura problem?
This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls via an analogue phone connected to the device through the SIP/IAX/Zap lines connected to the asterisk box. However - everytime the asterisk box does a 'sip reload' the Sipura 'blips' the phone within about 5-15 seconds of the reload. They've tried 3 different phones, but they all give a bleep or tinkle, so I guess the Sipura is pulsing the line whenever the asterisk box reloads the sip configuration. The Sipura appears to be at the latest and greatest firmware (according to my colleague at the far-end), but I've not had any first-hand experience of these boxes. It also appears to blip the phone seemingly randomly too - eg. in the middle of the night which isn't useful... (We could cope if it was jsut when we made configuration changes, but not seemingly at random - which I'm guessing is the SIP registration period timing out) Anyone seen anything like this before? I've had a look at a tshark dump on the line, but I'm not convinced there's actually anything wrong with it - maybe there's some setting wrong or not set in the sipura's config? Any clues welcome! Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quality problems with ISDN PRI
Hi, On Apr/23/2008, Steve Totaro wrote: On Tue, Apr 22, 2008 at 7:10 AM, Carles Pina i Estany [EMAIL PROTECTED] wrote: Hello, We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1 card, 3 SPANs configured and OK and one SPAN unconfigured. In our tests it works fine, but when it has a big laod of calls (say, from 40 to 60) we have quality problems: some calls has the sound cut-off (during the call, voice was not stable) The IRQ card is alone, CPU load was not high, network was fine for sure. This server is receiving the calls from SIP channels and routing to the primaries. It's a HP server, multicore, multiCPU. I'm wondering if someone has had these kind of problems (quality problems, sound cut off) with 40 and 60 calls but not with 2 or 3, using Digium cards. Bit later I will call to Digium but I thought that here there is lot of people with lot of experience with these cards. Thank you, Just curious, are you recording these calls because that is around the I/O threshold for audio issues when recording all calls. no, we are not recording calls. Load average is very empty. We are in contact with Spanish Digium partner... Also, you say no network issues but what is the rating of your switches PPS (often overlooked for speed such as 100mb or 1000mb)? 100 Mbps, enough for 50 - 60 calls Thanks, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided 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] changing of ssrc between early-media and call media
Greetings, upgrading from 1.4.17 to 1.4.19 some asterisk gateway of ours (used for gatewaying ISDN-PRI and SIP), I noticed an annoying thing: when the PSTN party answers, for a few seconds (4/5 sec typical) some SIP client could not hear anything (the ringing was heard well!), then the audio comes back again with no problem. Looking for any differences between the behaviour of version 1.4.17 and 1.4.19 I found that in the new version the RTP stream changes SSRC between the early media session and the actual call session. This seems to me quite pretty, and a major part of SIP clients seems not to be disturbed by it. Anyway I'd like to ask you a couple of things on this issue: 1) Is the changing of ssrc standard compliant? (I suppose yes, because the source changes from the Asterisk generating the ringing tone to the remote PSTN party actual speech, but I am not sure at all on this). 2) Do you know a way for avoiding such a change, in the meanwhile the SIP clients having problems will be appropriately patched? Maybe, I don't know, suggesting the PSTN to generate the ringing tone: how? Thanks, Francesco Castellano ___ -- Bandwidth and Colocation Provided 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] Quality problems with ISDN PRI
Also, you say no network issues but what is the rating of your switches PPS (often overlooked for speed such as 100mb or 1000mb)? 100 Mbps, enough for 50 - 60 calls Thanks, I asked for PPS (packets per second) not Mbps they are very different. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Polycom 3.0
Polycom is affiliated with the project in some way.. They also have an official Polycom moderated vendor forum. -Jon - Original Message - From: Andreas van dem Helge [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 How do they get away with that? On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html -Jon - Original Message - From: Darrick Hartman (lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 28, 2008 5:18:26 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 Andreas van dem Helge wrote: Anyone have a download link for 3.0 SIP firmware? If you are going to say ask polycom or ask your vendor don't even waste your time posting. I've asked the Nazis and they'll probably take 1 week. Suggest you get a different vendor then. I got a response from mine within a few hours. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided 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
[asterisk-users] RAS with Asterisk and PRI
Hi all, i have found the two possible solution for doing RAS with Asterisk: 1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD) I have downloaded the tgz proposed in the Wiki: app_pppd-060822.tgz But it do not compiler under Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q. First this #if gave me an error #if (ASTERISK_VERSION_NUM = 010400) #define ASTERISK_NEW_MODULE_INTERFACE #warning ASTERISK_NEW_MODULE_INTERFACE defined #endif [EMAIL PROTECTED]:/usr/src/app_pppd # make dep rm -f .depend touch .depend for i in *.c; do gcc -g -Wall -O -D_ISOC99_SOURCE -D_GNU_SOURCE -I/usr/include -M $i ;done .depend app_pppd.c:43:6: error: invalid digit 9 in octal constant But since this is only an test for asterisk version, i my version seems to fullfill the requirement, i just commented it out (only the diff, so that the #define will be used!) After this the make dep goes through, but make does not: [EMAIL PROTECTED]:/usr/src/app_pppd # make gcc -c -g -Wall -O -D_ISOC99_SOURCE -D_GNU_SOURCE -I/usr/include -o app_pppd.o app_pppd.c app_pppd.c:45:2: warning: #warning ASTERISK_NEW_MODULE_INTERFACE defined app_pppd.c:827: error: conflicting types for ‘unload_module’ /usr/include/asterisk/module.h:57: error: previous declaration of ‘unload_module’ was here app_pppd.c:840: error: conflicting types for ‘load_module’ /usr/include/asterisk/module.h:46: error: previous declaration of ‘load_module’ was here app_pppd.c:848: error: conflicting types for ‘description’ /usr/include/asterisk/module.h:75: error: previous declaration of ‘description’ was here app_pppd.c:861: error: conflicting types for ‘key’ /usr/include/asterisk/module.h:92: error: previous declaration of ‘key’ was here make: *** [app_pppd.o] Fehler 1 Does anyone know if app_pppd compile under Asterisk 1.2.9 ? I think this solution is quite attractive, since no zaptel patching is required ... 2) ZapRAS (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ZapRAS) Here one need to patch zaptel but the download link is broken: ftp://ftp.digium.com/pub/zaptel/misc/ Furthermore a Howto this all the steps would be very nice. For example the Wiki say this: - You need to apply the PPPoE patch then the Zaptel patch. Well, i have to work not very often with patch but i know that it matters what options you have to supply and the location from there you execute the command. After studying the Wiki page it seems that the patches only affect the ppp-package. Can someone explain to me why i have to re-compile zaptel? Do i have to change something there too? Why re-compile asterisk? No changes/patches in the asterisk source code are mentioned ... app_zapras.c is part of the asterisk packages, it is compiled and the .so file lies ready to use. Do i miss something that makes these recompiling necessary? After all the Link to the patches seems to be outdated: ftp://ftp.digium.com/pub/zaptel/misc/ should be changed to http://downloads.digium.com/pub/zaptel/misc/ I can do this myself the moment i got my password back for the Wiki ;) Thanks for any help/suggestions offered, Tobias Wolf ___ -- Bandwidth and Colocation Provided 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] tftp issue
Jerry Geis wrote: I have xinet tftp running on centos 5.1 It seems to be running on the local network eht0 fine. My box has 2 nics. however when I connect to eth1 for tftp I get: in.tftpd[5084]: tftpd: read(ack): Connection refused How can I get tftp working on BOTH eth0 and eth1 for my phone config files. man page for in.tftpd says it automatically runs for all local networks on port 69. Is eth1 not a local network? How do I get tftp to response on eth1? Thanks, Jerry Sorry - I got I first have to do yum install tftp-server this installs xinetd... Its still early Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TE420B (Four Port E1 PRI PCI-E x1 with Echo Cancellation), URGENT
Hello, I'm selling Digium TE420B (Four Port E1 PRI PCI-E x1 with Echo Cancellation), a brandly new one. The price is $1200. Urgent! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Polycom 3.0
An amazing change from the old days when you could only get firmware from a Polycom authorized distributer. Jonathan C. Bailey wrote: Polycom is affiliated with the project in some way.. They also have an official Polycom moderated vendor forum. -Jon - Original Message - From: Andreas van dem Helge [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 How do they get away with that? On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Polycom 3.0
Lol poorly moderated.post a question and then listen to crickets waiting for an answer. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, 29 April 2008 8:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Polycom 3.0 An amazing change from the old days when you could only get firmware from a Polycom authorized distributer. Jonathan C. Bailey wrote: Polycom is affiliated with the project in some way.. They also have an official Polycom moderated vendor forum. -Jon - Original Message - From: Andreas van dem Helge [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk- [EMAIL PROTECTED] Sent: Tuesday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 How do they get away with that? On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware- polycom-0-3.0.1-2.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc creating records or best practice
Robert,
You can access CDR information within the dialplan using the CDR
variable. I'm doing something very similar with a DISA feature for our
employees. We use ODBC to validate them against an existing MSSQL server
(check their employee ID pin number) then when all is well, I write
some information about the call (including the uniqueid field) out to a
'tracking' table I setup. Then I can join the tracking table and the cdr
table on the uniqueid column and associate employees with calls.
In my dialplan, I use the following snippet for setting the values in
the tracking table: (The DBNIS= line is where I do the insert)
exten = valid_login,1,NoOp()
exten = valid_login,n,Set(CALLDATE=${STRFTIME(${EPOCH},GMT+5,%x %X)})
exten = valid_login,n,Set(CLID=${CALLERID(num)})
exten = valid_login,n,Set(UNID=${CDR(uniqueid)})
exten = valid_login,n,Set(DBINS =
${ODBC_DISA(${CALLDATE},${CLID},${ID_ENTERED},${UNID})})
exten =
valid_login,n,Playback(/var/lib/asterisk/sounds/custom/disa_greet3)
exten = valid_login,n,DISA(no-password,from-disa,XXX
614)
exten = valid_login,n(end),Hangup
HTH!
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
McNaught
Sent: Monday, April 28, 2008 6:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] func_odbc creating records or best practice
Hi,
I am trying to write a custom application which will integrate with an
existing MSSQL crm system.
We need to get ahold of the CDR(uniqueid) field in during call-time -
I see from doing a DumpChan(), the CDR unique ID is available as soon
as the call is created. CDRs usind odbc are only written once the
call is completed. Does anyone know if it is possible to use
func_odbc to create a temporary record then delete it so that this
information is available to MSSQL. I was not sure if func_odbc was
limited to just using UPDATE/SELECT queries.
Would there be a better way to do this using the AMI or AGI? It just
seems a little strange to use a database for storing temporary data
such as this?
Thanks in Advance
Robert McNaught
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Re: [asterisk-users] RAS with Asterisk and PRI
On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote: Hi all, i have found the two possible solution for doing RAS with Asterisk: 1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD) I have downloaded the tgz proposed in the Wiki: app_pppd-060822.tgz But it do not compiler under Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q. First this #if gave me an error #if (ASTERISK_VERSION_NUM = 010400) #define ASTERISK_NEW_MODULE_INTERFACE #warning ASTERISK_NEW_MODULE_INTERFACE defined #endif This is for Asterisk 1.4 . There is bristuff for 1.4 (bristuff 0.4.x). Or, alternatively, backport it yourself... -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need a monitor for asterisk
Hello: I have asterisk configured to use sip with two providers. Checking with the command sip show registry I found that sometimes is not registered. ?Is it there anyway to configure asterisk to restablish the connection with the providers automatically? Thanks in advance for any answer. Enediel _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided 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] Newbie Queue: greetings when first joiningqueue
Lee, John (Sydney) wrote:
Check the number of calls waiting in the queue, then play the message
if
more than 0
example code (written in the TBird IDE)
Exten = 100,1,Answer()
Exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
Exten = 100,n,GotoIf($[${NumWaiting} = 0]?JoinQueue)
Exten = 100,n,PlayBack(MyMessage)
Exten = 100,n(JoinQueue),Queue(MyQueueName)
Exten = 100,n,Hangup()
So, if there are no members in the queue, jump directly to the queue
application, otherwise play the message first.
Thanks Julian and it certainly works.
I have got another question if I may.
If there is just one agent in the queue and he put on Do-Not-Disturb,
certainly in this case the queue count will be zero but I would still
like Asterisk to play back Welcome to XYZ, your call is important to us
... please stay on the line, the logic above would fail to play back
the intro message.
I thought about trapping DIALSTATUS but if there is actually no dial
cmd, how can I trap the DND then and play back the message again?
Any thoughts?
The only possibility I can see would be the QUEUE_MEMBER_COUNT function,
however if the agent is using the DND feature of their phone, this is
very unlikely to work. The only other method I can think of would be to
call the queue using the n option to enter the queue and when the
Queue application returns (which it will do if the call hasn't been
answered on the first try - either due to a timeout or because the agent
is in DND mode) to play the announcement to the caller.
Try something like...
exten = 100,1,Answer()
exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
exten = 100,n,GotoIf($[${NumWaiting} = 0]?firstcaller)
exten = 100,n,Goto(announce)
exten = 100,n(firstcaller),Queue(MyQueueName,nr)
exten = 100,n(announce),PlayBack(MyMessage)
exten = 100,n,Queue(MyQueueName)
exten = 100,n,Hangup()
Calling the queue with the options nr means the queue will play the
ring tone (omit the r if this isn't the desired behaviour) and drop
through to the next statement if the call times out to all available
agents... which would then play the announcement and put them back in
the queue.
Theoretically, anyway. :)
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Re: [asterisk-users] Hyperthreading and multicore
This is my understanding of hyper threading, which I believe to be accurate. Basically, as some have mentioned previously, the OS 'sees' your single physical core processor as 2 logical processors, in generally, logical processors are treated exactly as if they were real processors, and in the case of many OS's. they probably don't understand the difference - Linux however does have specific SMT support for hyperthreaded cores. Basically not all CPU instructions take the same amount of clock cycles to complete, some may take 3, some may take 7, etc. Many of these clock cycles actually goto waste because the CPU is waiting for something, for example, an instruction that involves a fetch from memory, if this takes 7 clock cycles to complete, 4 of those cycles might go wasted while the CPU essentially just sits there and waits for the data to be fetched form RAM, L1 or L2, or L3 cache. Hyperthreading essentially puts these wasted CPU cycles to use by allowing the CPU to execute a separate thread while it would otherwise be idle waiting. To me Hyperthreading is an excellent technology... I;m all about efficiency and trying to maximize resource usage whenever possible... and that exactly what hyper threading does. That all being said... Hyper threading should not be thought of as effectively doubling your CPU power... as previous posters have said, Hyper threading will result in single threaded applications actually running slower.. this is because you still have other background processes running which may run on the other logical processor which could steal CPU cycles away from your main application... since you essentially have 2 threads executing on the same physical core, there are going to be times when one thread has to wait extra clock cycles while the other thread is executing. Remember its only those normally wasted clock cycles that you are going to gain a performance boost out of by making use of them... only 1 thread can actually be executing at any given time, so the CPU has to schedule these and try balance the threads equally so they each get an equal share of the physical core. I can't say how Asterisk behaves or makes use of additional cores or if hyper threading is advantageous to Asterisk or not... I don't know enough about the low level parts of Asterisk enough to make an informed opinion about that. I just thought I'd throw in my 2 cents about what hyper threading is and what it does. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis Sent: Tuesday, April 29, 2008 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hyperthreading and multicore Matt Florell wrote: Also, I have heard HT processors explained this way, on an HT processor it's like running 2 virtual processors at 70% of the specs of the processor with HT turned off. It's not really like that in all situations, but overall it has held pretty much true for me in most non-Asterisk situations. Asterisk didn't benefit much from having HT enabled on a P4 with HT capability. That wouldn't surprise me - after all, HyperThreading works on the principle of allowing two threads to use different dedicated processor resources (such as floating point math processors and so on) at the same time... however if two threads are trying to use the same processor resource, one thread will be suspended until that resource becomes available. ___ -- Bandwidth and Colocation Provided 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] Outbound international calls over BT ISDN30
Hello all As always I'm trying the mailing list as a last resort as I'm out of options. I am seemingly unable to dial international numbers over our BT ISDN30 line. I've checked with BT and the number format they're expecting is: 00CCnumber (where CC is the country code). But this doesn't work. Looking at the PRI debug, the most notable error seems to be: Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] I've also tried different number formats, including: +CCnumber +0CCnumber But to no avail. Anybody know what I'm doing wrong? Here's a complete debug dump of a failed international call to the US: Many thanks Stu -- Executing [EMAIL PROTECTED]:2] Dial(SIP/sbf-b7c104e0, Zap/g1/0012127551200) in new stack -- Making new call for cr 33090 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 322/0x142) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] [6c 0d 21 80 30 31 36 31 34 38 36 37 37 38 30] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '01614867780' ] [70 0e a1 30 30 31 32 31 32 37 35 35 31 32 30 30] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0012127551200' ] [a1]on*CLI Sending Complete (len= 1) q931.c:2881 q931_setup: call 33090 on channel 3 enters state 1 (Call Initiated) -- Called g1/0012127551200 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3428 q931_receive: call 33090 on channel 3 enters state 3 (Outgoing call Proceeding) -- Zap/3-1 is proceeding passing it to SIP/sbf-b7c104e0 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) q931.c:3563 q931_receive: call 33090 on channel 3 enters state 12 (Disconnect Indication) -- Channel 0/3, span 1 got hangup request, cause 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2716 q931_release: call 33090 on channel 3 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 322/0x142) (Originator) Message type: RELEASE (77) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Hangup(SIP/sbf-b7c104e0, ) in new stack == Spawn extension (macro-outgoing, s, 3) exited non-zero on 'SIP/sbf-b7c104e0' in macro 'outgoing' == Spawn extension (macro-outgoing, s, 3) exited non-zero on 'SIP/sbf-b7c104e0' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3503 q931_receive: call 33090 on channel 3 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call,
[asterisk-users] Debugging DTMF
Hi All, I'm trying to debug DTMF issues I have with certain endpoint conferencing systems (external, 3rd party). On our A*k server I log DTMF, and I see that coming through in the log. What I'd like to see is what is sent onto our VoIP carrier over SIP. I can do a tcpdump of the packets, but what am I then looking for? Would it be in the RTP audio stream or within the SIP protocol?? I'm using Wireshark to decode... Thanks, Adrian ___ -- Bandwidth and Colocation Provided 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] Outbound international calls over BT ISDN30
You might need to set the dialplan to international or so in the config files. Zoa Stuart Ford wrote: Hello all As always I'm trying the mailing list as a last resort as I'm out of options. I am seemingly unable to dial international numbers over our BT ISDN30 line. I've checked with BT and the number format they're expecting is: 00CCnumber (where CC is the country code). But this doesn't work. Looking at the PRI debug, the most notable error seems to be: Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] I've also tried different number formats, including: +CCnumber +0CCnumber But to no avail. Anybody know what I'm doing wrong? Here's a complete debug dump of a failed international call to the US: Many thanks Stu -- Executing [EMAIL PROTECTED]:2] Dial(SIP/sbf-b7c104e0, Zap/g1/0012127551200) in new stack -- Making new call for cr 33090 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 322/0x142) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] [6c 0d 21 80 30 31 36 31 34 38 36 37 37 38 30] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '01614867780' ] [70 0e a1 30 30 31 32 31 32 37 35 35 31 32 30 30] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0012127551200' ] [a1]on*CLI Sending Complete (len= 1) q931.c:2881 q931_setup: call 33090 on channel 3 enters state 1 (Call Initiated) -- Called g1/0012127551200 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3428 q931_receive: call 33090 on channel 3 enters state 3 (Outgoing call Proceeding) -- Zap/3-1 is proceeding passing it to SIP/sbf-b7c104e0 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) q931.c:3563 q931_receive: call 33090 on channel 3 enters state 12 (Disconnect Indication) -- Channel 0/3, span 1 got hangup request, cause 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2716 q931_release: call 33090 on channel 3 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 322/0x142) (Originator) Message type: RELEASE (77) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Hangup(SIP/sbf-b7c104e0, ) in new stack == Spawn extension (macro-outgoing, s, 3) exited non-zero on 'SIP/sbf-b7c104e0' in macro 'outgoing' == Spawn extension (macro-outgoing, s, 3) exited non-zero on 'SIP/sbf-b7c104e0' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 322/0x142) (Terminator) Message type:
Re: [asterisk-users] RAS with Asterisk and PRI
Many people think ZapRAS is for modem dialin. None of the RAS stuff support modems, as far as I know. The RAS stuff in Asterisk is for networking via ISDN, rather than modem. Tzafrir Cohen wrote: On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote: Hi all, i have found the two possible solution for doing RAS with Asterisk: 1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD) I have downloaded the tgz proposed in the Wiki: app_pppd-060822.tgz But it do not compiler under Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q. First this #if gave me an error #if (ASTERISK_VERSION_NUM = 010400) #define ASTERISK_NEW_MODULE_INTERFACE #warning ASTERISK_NEW_MODULE_INTERFACE defined #endif This is for Asterisk 1.4 . There is bristuff for 1.4 (bristuff 0.4.x). Or, alternatively, backport it yourself... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided 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] Outbound international calls over BT ISDN30
See below In article [EMAIL PROTECTED], Stuart Ford [EMAIL PROTECTED] wrote: Hello all As always I'm trying the mailing list as a last resort as I'm out of options. I am seemingly unable to dial international numbers over our BT ISDN30 line. I've checked with BT and the number format they're expecting is: 00CCnumber (where CC is the country code). But this doesn't work. Looking at the PRI debug, the most notable error seems to be: Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] I've also tried different number formats, including: +CCnumber +0CCnumber But to no avail. Anybody know what I'm doing wrong? Here's a complete debug dump of a failed international call to the US: Many thanks Stu -- Executing [EMAIL PROTECTED]:2] Dial(SIP/sbf-b7c104e0, Zap/g1/0012127551200) in new stack -- Making new call for cr 33090 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 322/0x142) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 83] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 3 ] [6c 0d 21 80 30 31 36 31 34 38 36 37 37 38 30] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '01614867780' ] [70 0e a1 30 30 31 32 31 32 37 35 35 31 32 30 30] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0012127551200' ] I think this is your problem. It is using National as the Type Of Number, but you have given it an international number. [a1]on*CLI Sending Complete (len= 1) What values do you have in zapata.conf for pridialplan, internationalprefix, nationalprefix and localprefix? Try each of the following two sets of parameters: (A) pridialplan=dynamic internationalprefix=00 nationalprefix=0 localprefix= (B) pridialplan=unknown internationalprefix= nationalprefix= localprefix= I normally use the first set. In your case it would then send the destination number as '12127551200' with a TON of International. Dialling a UK number 01234 567890 would be sent as '1234567890' with a TON of National. The second set of parameters sends all numbers as-is, but with a TON of Unknown. This may allow the exchange to do normal interpretation like it does with analogue phones, i.e. 00CC for international and 0[1-9]... for national. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - 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
[asterisk-users] Odd Zaptel Issue - Strange State 6?
Hello! As the subject states, I'm experiencing an odd issue with one of my client's systems. We're running a Sangoma A400D card with 8 FXO channels. The POTS line on channel 5, whenever called, generates the following message on the asterisk console: Apr 29 10:35:08 WARNING[16927]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 5 Apr 29 10:35:09 WARNING[16927]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 5 Eventually, the call will be answered and passed to the inbound context (a ring group). When anyone answers the call, all they hear is silence. The calling party that called in on ZAP/5 still hears the line ringing. It is very odd. We're using Asterisk 1.2.12.1, Zaptel 1.2.12, and Wanpipe 3.2.1 on CentOS 4.4. Thank you for your help!!! Tim Nelson Systems/Network Support Rockbochs Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound international calls over BT ISDN30
2008/4/29 Tony Mountifield [EMAIL PROTECTED]: [snip] What values do you have in zapata.conf for pridialplan, internationalprefix, nationalprefix and localprefix? Try each of the following two sets of parameters: (A) pridialplan=dynamic internationalprefix=00 nationalprefix=0 localprefix= (B) pridialplan=unknown internationalprefix= nationalprefix= localprefix= I'll just add that while (A) should be more correct, (B) seems to work with a larger number of ISDN30 systems in the UK. Regards, Steve ___ -- Bandwidth and Colocation Provided 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] Outbound international calls over BT ISDN30
On Tue, Apr 29, 2008 at 05:00:26PM +0100, Steve Davies wrote: 2008/4/29 Tony Mountifield [EMAIL PROTECTED]: [snip] What values do you have in zapata.conf for pridialplan, internationalprefix, nationalprefix and localprefix? Try each of the following two sets of parameters: (A) pridialplan=dynamic internationalprefix=00 nationalprefix=0 localprefix= (B) pridialplan=unknown internationalprefix= nationalprefix= localprefix= I'll just add that while (A) should be more correct, (B) seems to work with a larger number of ISDN30 systems in the UK. In fact, is there any reason not to make 'unknown' the default of pridialplan? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RAS with Asterisk and PRI
Eric Wieling schrieb: Many people think ZapRAS is for modem dialin. None of the RAS stuff support modems, as far as I know. The RAS stuff in Asterisk is for networking via ISDN, rather than modem. This is exactly what we want ZapRAS to use for ;) Maybe anyone can enlighten me in answering my questions according ZapRAS, or are they too trivial ;) Have i overlooked some useful source of information ? Tzafrir Cohen wrote: On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote: Hi all, i have found the two possible solution for doing RAS with Asterisk: 1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD) I have downloaded the tgz proposed in the Wiki: app_pppd-060822.tgz But it do not compiler under Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q. First this #if gave me an error #if (ASTERISK_VERSION_NUM = 010400) #define ASTERISK_NEW_MODULE_INTERFACE #warning ASTERISK_NEW_MODULE_INTERFACE defined #endif This is for Asterisk 1.4 . There is bristuff for 1.4 (bristuff 0.4.x). Or, alternatively, backport it yourself... -- Tobias Wolf Leiter Softwareentwicklung / Kommunikationslösungen Evision GmbH Wittekindstr. 105 44139 Dortmund Tel: +49 (0)231 - 47790 307 Fax: +49 (0)231 - 47790 500 http://www.evision.de This electronic mail transmission and any accompanying attachments contain confidential information intended only for the use of the individual or entity named above. Any dissemination, distribution, copying or action taken in reliance on the contents of this communication by anyone other than the intended recipient is strictly prohibited. If you have received this communication in error please immediately delete the E-mail and notify the sender at the above E-mail address. Thank you. Hövener Trapp Evision GmbH, Dortmund - HRB Nr.12477, Registergericht Dortmund - Geschäftsführer Christoph Begall ___ -- Bandwidth and Colocation Provided 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] RAS with Asterisk and PRI
Alexander Lopez schrieb: If you want to do RAS (modem or data) I would suggest going with a Portmaster PM3(Livingston, then purchased by Lucent, They are reliable and pretty cheap. You can use Asterisk to route the calls into it if you use a two port card. You can find them at www.portmasters.com pm-3a-2t-r PM-3A-2T w/48 56k v.90 modems $395.00 $400.00 it will save you more in time and aggravation. Another solution that i thought of, was just to use an ordinary ISDN Card like an AVM!Fritz. Configure the pppd for this card, plug an cable from the Fritz to my Junghans Card. This is not very elegant, but cheap and saves one the hazzle with patching software that runs smoothly or has to be upgraded ... regards -- Tobias Wolf Leiter Softwareentwicklung / Kommunikationslösungen Evision GmbH Wittekindstr. 105 44139 Dortmund Tel: +49 (0)231 - 47790 307 Fax: +49 (0)231 - 47790 500 http://www.evision.de This electronic mail transmission and any accompanying attachments contain confidential information intended only for the use of the individual or entity named above. Any dissemination, distribution, copying or action taken in reliance on the contents of this communication by anyone other than the intended recipient is strictly prohibited. If you have received this communication in error please immediately delete the E-mail and notify the sender at the above E-mail address. Thank you. Hövener Trapp Evision GmbH, Dortmund - HRB Nr.12477, Registergericht Dortmund - Geschäftsführer Christoph Begall ___ -- Bandwidth and Colocation Provided 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] RAS with Asterisk and PRI
Tzafrir Cohen schrieb: On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote: Hi all, i have found the two possible solution for doing RAS with Asterisk: 1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD) I have downloaded the tgz proposed in the Wiki: app_pppd-060822.tgz But it do not compiler under Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q. First this #if gave me an error #if (ASTERISK_VERSION_NUM = 010400) #define ASTERISK_NEW_MODULE_INTERFACE #warning ASTERISK_NEW_MODULE_INTERFACE defined #endif This is for Asterisk 1.4 . There is bristuff for 1.4 (bristuff 0.4.x). Or, alternatively, backport it yourself... Ahh, thanks ... good to know ... i misinterpreted the source code as 1.04 and not as 1.4. maybe i will take a go at the bristuff for asterisk 1.4 regards, -- Tobias Wolf Leiter Softwareentwicklung / Kommunikationslösungen Evision GmbH Wittekindstr. 105 44139 Dortmund Tel: +49 (0)231 - 47790 307 Fax: +49 (0)231 - 47790 500 http://www.evision.de This electronic mail transmission and any accompanying attachments contain confidential information intended only for the use of the individual or entity named above. Any dissemination, distribution, copying or action taken in reliance on the contents of this communication by anyone other than the intended recipient is strictly prohibited. If you have received this communication in error please immediately delete the E-mail and notify the sender at the above E-mail address. Thank you. Hövener Trapp Evision GmbH, Dortmund - HRB Nr.12477, Registergericht Dortmund - Geschäftsführer Christoph Begall ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Polycom 3.0
The thing is that's still the case. If they really wanted change they'd post the newest version of the firmware at www.polycom.com which they dont (well technically yes but you need to be a member of their reseller crap) The byproduct of the corporate bureaucracy,. Isn't it great? On Tue, Apr 29, 2008 at 8:54 AM, Eric Wieling [EMAIL PROTECTED] wrote: An amazing change from the old days when you could only get firmware from a Polycom authorized distributer. Jonathan C. Bailey wrote: Polycom is affiliated with the project in some way.. They also have an official Polycom moderated vendor forum. -Jon - Original Message - From: Andreas van dem Helge [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 How do they get away with that? On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AJAM event subscription - Was: func_odbc creating records or best practice
Thanks guys.
I understand the ODBC and channel variables using Asterisk. However,
I am really looking for an ability for a webserver to subscribe to
channel status in asterisk and be informed when a call comes in and
show the callerID in real-time.
From the AMI, there is Newcallerid Event which would be suitable for
this purpose. I have been reading the voip-info pages and have set up
AJAM and can get results from doing http requests to the asterisk
server, however, this is in the form of an action, such as login,
rather than subscribing to an event.
I have been looking here for information
http://www.voip-info.org/wiki/view/asterisk+manager+events
Does anyone know if subscriptions is possible with AJAM?
Thanks
Robert
On Tue, Apr 29, 2008 at 6:14 AM, Gleim, Jason [EMAIL PROTECTED] wrote:
Robert,
You can access CDR information within the dialplan using the CDR
variable. I'm doing something very similar with a DISA feature for our
employees. We use ODBC to validate them against an existing MSSQL server
(check their employee ID pin number) then when all is well, I write
some information about the call (including the uniqueid field) out to a
'tracking' table I setup. Then I can join the tracking table and the cdr
table on the uniqueid column and associate employees with calls.
In my dialplan, I use the following snippet for setting the values in
the tracking table: (The DBNIS= line is where I do the insert)
exten = valid_login,1,NoOp()
exten = valid_login,n,Set(CALLDATE=${STRFTIME(${EPOCH},GMT+5,%x %X)})
exten = valid_login,n,Set(CLID=${CALLERID(num)})
exten = valid_login,n,Set(UNID=${CDR(uniqueid)})
exten = valid_login,n,Set(DBINS =
${ODBC_DISA(${CALLDATE},${CLID},${ID_ENTERED},${UNID})})
exten =
valid_login,n,Playback(/var/lib/asterisk/sounds/custom/disa_greet3)
exten = valid_login,n,DISA(no-password,from-disa,XXX
614)
exten = valid_login,n(end),Hangup
HTH!
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
McNaught
Sent: Monday, April 28, 2008 6:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] func_odbc creating records or best practice
Hi,
I am trying to write a custom application which will integrate with an
existing MSSQL crm system.
We need to get ahold of the CDR(uniqueid) field in during call-time -
I see from doing a DumpChan(), the CDR unique ID is available as soon
as the call is created. CDRs usind odbc are only written once the
call is completed. Does anyone know if it is possible to use
func_odbc to create a temporary record then delete it so that this
information is available to MSSQL. I was not sure if func_odbc was
limited to just using UPDATE/SELECT queries.
Would there be a better way to do this using the AMI or AGI? It just
seems a little strange to use a database for storing temporary data
such as this?
Thanks in Advance
Robert McNaught
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Re: [asterisk-users] realtime queue callers
--- Atis Lezdins [EMAIL PROTECTED] wrote: So, I suppose if MySQL dies in middle of operation, SELECT should fail and Asterisk should just continue with what it has in memory. Btw, You should be able to also use static or dynamic queue members (not realtime) in combination with realtime queue calls. That's what I would like to do: use dynamic queue members but rely on mysql for monitoring active queue calls. Btw, I never experienced that MySQL dies, it's more often that Asterisk dies. I agree. But I did have a strange case at one point. Would like to reduce point of failures anyway, as much as possible. So, are You interested in applying this patch yourself? I just wrote a simple AMI script which parses the output of show queues and sends relevant data to my custom application via sockets. The only problem is that I need to periodically run the script (cron) so it's rather inefficient. Maybe I could trigger the script on particular Manager events (such as run the script which parses 'show queues' only when I receive Agent* events). I don't want you to make the effort of finding that patch (as it seems you don't have it at hand now) if I may not need it. However, I think that your patch should hit SVN and I wouldn't mind testing it. Thanks, Vieri Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided 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] ooh323 asterisk 1.2.x
Hi, im trying to config ooh323 in asterisk. i compiled the one inside the asterisk-addons im trying to connect my softphone(Xmeeting). here's my ooh323.conf [marq] type=friend context=avaya ip=dynamic port=1720 e164=888 username=marq secret=marq -- Regards, Mark Quitoriano http://asterisk.org.ph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have pricing on the Color Polycom Phone?
On Tue, Apr 29, 2008 at 1:02 AM, Patrick [EMAIL PROTECTED] wrote: On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote: Anyone seen anything on the IP670 the Color Expansion? Great timing. Yesterday I was looking at the IP650 and wondered when the successor to the IP650 would arrive. Do you have a link or more info about the IP670? Thanks, Patrick No other infojust saw the link on Polycom's site. If you click the link, you get a 404. Will post info if I find it. -Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - CRM Integration
Is Sugar CRM the best Free CRM to be integrated with Asterisk ? Is Asterisk VoiceRD Integration the best integration patch to be used with Sugar CRM ? Is any other ? Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - CRM Integration
Is Sugar CRM the best Free CRM to be integrated with Asterisk ? Is Asterisk VoiceRD Integration the best integration patch to be used with Sugar CRM ? Is any other ? Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - CRM Integration
On 18:16, Tue 29 Apr 08, Fernando Berretta wrote: Is Sugar CRM the best Free CRM to be integrated with Asterisk ? Is Asterisk VoiceRD Integration the best integration patch to be used with Sugar CRM ? Is any other ? Have a look at Covide: http://sourceforge.net/projects/covide /shameless_plug -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - CRM Integration
vicidial ... vicidialNOW ___ -- Bandwidth and Colocation Provided 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] Jitter buffer not used in SIP - chan_local - ZAP path even with /nj for local channels
Hi, Asterisk 1.4 Working (jitter buffers created as expected): ZAP - SIP SIP - ZAP Not working (no jitter buffers created): SIP - chan_local (with /nj) - ZAP SIP - chan_local (with /j) - ZAP SIP - chan_local (with no flags) - ZAP I have this in zapata.conf: jbenable=yes jbforce=no jbimpl=fixed jbmaxsize=300 Is there something I haven't tried that will make this work or will I have to change my dialplan so it doesn't use local channels? Thanks, Mike PS, here are some pages that I have used as sources of information: No mention of /j for local channels http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels Nothing about local channels http://www.asterisk.org/node/48317 Mentions /j for local channels to apps http://www.voip-info.org/tiki-index.php?page=Asterisk+new+jitterbuffer ___ -- Bandwidth and Colocation Provided 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] Sending caller name out PRI?
I have a PRI connected to a traditional PBX using NI-2 and a typical config (further below). When I call from a SIP/IAX phone to an extension on the PBX, only the number makes it through. If I plug that same port on the PBX to a carrier the PBX presents both name and number. Hints or pokes to relevant chapters in documentation? My config is essentially like one found here: http://www.voip-info.org/files/nortel-asterisk-0.2.pdf The author makes no reference to CNID, so I'm assuming that he wasn't bothered by it not working. Ideas? The system is a trixbox 2.6.0.7 which includes zaptel-1.4.9.2-8. ... -- Executing [EMAIL PROTECTED]:22] NoOp(SIP/3991-b7900488, CallerID set to Peter NPANXX3991) in ne Executed application: Noop Executed application: Macro -- Executing [EMAIL PROTECTED]:12] AGI(SIP/3991-b7900488, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix == Begin MixMonitor Recording SIP/3991-b7900488 -- AGI Script fixlocalprefix completed, returning 0 Executed application: AGI -- Executing [EMAIL PROTECTED]:13] Set(SIP/3991-b7900488, OUTNUM=4342) in new stack Executed application: Set -- Executing [EMAIL PROTECTED]:14] Set(SIP/3991-b7900488, custom=ZAP/g14) in new stack Executed application: Set -- Executing [EMAIL PROTECTED]:15] GotoIf(SIP/3991-b7900488, 1?gocall) in new stack -- Goto (macro-dialout-trunk,s,17) Executed application: GotoIf -- Executing [EMAIL PROTECTED]:17] Macro(SIP/3991-b7900488, dialout-trunk-predial-hook|) in new stack Context 'macro-dialout-trunk-predial-hook' for macro 'dialout-trunk-predial-hook' lacks 's' extension, priority 1 Executed application: Macro -- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/3991-b7900488, 0?bypass|1) in new stack Executed application: GotoIf -- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/3991-b7900488, 0?customtrunk) in new stack Executed application: GotoIf -- Executing [EMAIL PROTECTED]:20] Dial(SIP/3991-b7900488, ZAP/g14/4342|300|) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g14/4342 Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 4 -- Zap/73-1 is proceeding passing it to SIP/3991-b7900488 -- Zap/73-1 is ringing Echo cancellation already on -- Zap/73-1 answered SIP/3991-b7900488 ... zapata.conf (includes inline and comments purged) Spans 3 and 4 connect to the PBX. (Yes the restating of the defaults are redundant, but I'm willing to try any goofiness to make it work.) --- [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming group=0,11 context=from-carrier-custom callerid=asreceived usecallerid=yes hidecallerid=no switchtype = national signalling = pri_cpe pridialplan=national prilocaldialplan=national channel = 1-23 group= context=default ; Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 group=0,12 context=from-carrier-custom callerid=asreceived usecallerid=yes hidecallerid=noswitchtype = national signalling = pri_cpe pridialplan=national prilocaldialplan=national channel = 25-47 group= context=default ; Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 group=0,13 context=from-nortel-custom callerid=asreceived usecallerid=yes hidecallerid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=incoming switchtype = national signalling = pri_net pridialplan=national prilocaldialplan=national channel = 49-71 group= context=default ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=0,14 context=from-nortel-custom callerid=asreceived usecallerid=yes hidecallerid=no switchtype = national signalling = pri_net pridialplan=national prilocaldialplan=national channel = 73-95 group= context=default group=1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk sends 486 Busy Here instead of 600Busy Everywhere
Thanks Olle and Jared for your reply. This clears a lot of my doubts. Thanks once again. Regards, Aadil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E Sent: Wednesday, April 23, 2008 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: 'Vinod Kumar Singh' Subject: Re: [asterisk-users] Asterisk sends 486 Busy Here instead of 600Busy Everywhere 22 apr 2008 kl. 14.41 skrev Aadilkhan Maniyar: Hi, We have a scenario wherein the endpoint needs to send a 600 Busy Everywhere after receiving an INVITE. I am using SIPp as this end point. SIPp is configured as UE2. Now when UE1 calls UE2 (SIPp) receives the INVITE and responds with a 600 Busy Everywhere. But when Asterisk receives this 600 response it sends out a 486 Busy Here to UE1. Ideally Asterisk should be relaying the 600 response. What I fail to get is, why does Asterisk need to send 486 instead of 600. Is there any configuration that needs to be done in order to achieve this or is this a default behavior of Asterisk. I am using Asterisk 1.4.17. As Jared said, we're a multiprotocol PBX. When we receive an error code in a signalling channel - a channel driver - we have to translate all those codes into some sort of esperanto that we handle in the sip core. We might have a call that forks to both IAX2, ZAP and SIP and need to handle error codes from all those protocols, so we translate everything into ISDN cause codes. Now, ISDN haven't got any difference between local busy and busy everywhere. So when we translate back, we pick the 486 code. I hope you now understand why we always send out the same SIP error code as get on the outbound channel. We do follow IETF specifications for the translations. Regards, /Olle ___ -- Bandwidth and Colocation Provided 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
