Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9
Cyprus VoIP wrote: Thank you for your answer. The 'internal extension' is indeed a T.38 capable device that works perfectly when connected directly to the Proxy/ITSP. As you said, the key to debugging/resolving this issue is the logger. I wasn't aware of this file. this is what I have there: ... ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error ;full = notice,warning,error,debug,verbose ... Should I change the console... line or uncomment the ;full... line? Either one is fine; using 'full' is actually a bit better, because the color highlighting done on the console sometimes makes console captures hard to read. Hi, So, I enabled the full logger, and the strange thing I see is this message: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session It seems that this might be the reason Asterisk initiates a reINVITE with voice codecs, after connecting the 2 parties. Is there a way to disable that action, or do we need to add T.38 somehow to the list of codecs? I followed the instructions on the default sip.conf to include the line t38pt_udptl=yes,redundancy in the general section and in each of the parties. Thanks. ___ -- 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] hey please help me my 3rd email of how to change From fileld username in sip packet
Dear Hakan, thank you for your information on this issue it does change but it only changed in REQUEST URI field not in From Field, Date: Fri, 4 Dec 2009 11:32:59 +0500 From: Masood Ahmed masoo...@gmail.com Subject: [asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet To: asterisk-users@lists.digium.com Message-ID: 1fda62f80912032232o56fe6969se02526bf4ccf1...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as asterisk and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 - IP2:5060 INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP IP1:5060;branch=z9hG4bK- 966123148--16781 75694--693700493-4-..Via: SIP/2.0/UDP IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57 804d...@ip1..to 804d...@117.20.20.234..to: sip:3347871...@ip1.. * *From: asterisksip:aster...@ip1:5065;tag=as0cae0b** see the last part this is what that i want to change here in from it should be some CLI thanks Masood -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091204/9a953328/attachment-0002.htm -- Message: 24 Date: Fri, 4 Dec 2009 11:32:59 +0500 From: Masood Ahmed masoo...@gmail.com Subject: [asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet To: asterisk-users@lists.digium.com Message-ID: 1fda62f80912032232o56fe6969se02526bf4ccf1...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as asterisk and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 - IP2:5060 INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP IP1:5060;branch=z9hG4bK- 966123148--16781 75694--693700493-4-..Via: SIP/2.0/UDP IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57 804d...@ip1..to 804d...@117.20.20.234..to: sip:3347871...@ip1.. * *From: asterisksip:aster...@ip1:5065;tag=as0cae0b** see the last part this is what that i want to change here in from it should be some CLI thanks Masood -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091204/9a953328/attachment-0003.htm -- Message: 25 Date: Fri, 4 Dec 2009 09:29:35 +0200 From: Hakan C ella4e...@gmail.com Subject: Re: [asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: ef4e56e70912032329m659f0b89p8946f3c96c2c8...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 exten = 111,1,Set(CallerID(num)=123456) On Fri, Dec 4, 2009 at 8:32 AM, Masood Ahmed masoo...@gmail.com wrote: hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as asterisk and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 - IP2:5060 INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP IP1:5060;branch=z9hG4bK- 966123148--16781 75694--693700493-4-..Via: SIP/2.0/UDP IP1:5065;branch=z9hG4bK00749b6d;rport
Re: [asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet
exten = 111,1,Set(CallerID(name)=MyPBX) or exten = 111,1,Set(CallerID(all)=123456) On 12/4/09, Masood Ahmed masoo...@gmail.com wrote: Dear Hakan, thank you for your information on this issue it does change but it only changed in REQUEST URI field not in From Field, Date: Fri, 4 Dec 2009 11:32:59 +0500 From: Masood Ahmed masoo...@gmail.com Subject: [asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet To: asterisk-users@lists.digium.com Message-ID: 1fda62f80912032232o56fe6969se02526bf4ccf1...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as asterisk and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 - IP2:5060 INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP IP1:5060;branch=z9hG4bK- 966123148--16781 75694--693700493-4-..Via: SIP/2.0/UDP IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57 804d...@ip1..to 804d...@117.20.20.234..to: sip:3347871...@ip1.. * *From: asterisksip:aster...@ip1:5065;tag=as0cae0b** see the last part this is what that i want to change here in from it should be some CLI thanks Masood -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091204/9a953328/attachment-0002.htm -- Message: 24 Date: Fri, 4 Dec 2009 11:32:59 +0500 From: Masood Ahmed masoo...@gmail.com Subject: [asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet To: asterisk-users@lists.digium.com Message-ID: 1fda62f80912032232o56fe6969se02526bf4ccf1...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as asterisk and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 - IP2:5060 INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP IP1:5060;branch=z9hG4bK- 966123148--16781 75694--693700493-4-..Via: SIP/2.0/UDP IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57 804d...@ip1..to 804d...@117.20.20.234..to: sip:3347871...@ip1.. * *From: asterisksip:aster...@ip1:5065;tag=as0cae0b** see the last part this is what that i want to change here in from it should be some CLI thanks Masood -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091204/9a953328/attachment-0003.htm -- Message: 25 Date: Fri, 4 Dec 2009 09:29:35 +0200 From: Hakan C ella4e...@gmail.com Subject: Re: [asterisk-users] hey please help me my 3rd email of how to change From fileld username in sip packet To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: ef4e56e70912032329m659f0b89p8946f3c96c2c8...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 exten = 111,1,Set(CallerID(num)=123456) On Fri, Dec 4, 2009 at 8:32 AM, Masood Ahmed masoo...@gmail.com wrote: hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as asterisk and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 - IP2:5060 INVITE sip:0423347871...@ip2:5060 SIP
Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9
Set 'canreinvite=no' on all applicable peers? I tried with yes and no. No difference. I'm almost certain it's related to the Keeping RTP active during T.38 session issue. ___ -- 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] Dahdi and Junghanns QuadBRI
2009/11/19 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Nov 19, 2009 at 07:01:13AM +0100, Olivier wrote: Hi, I'm using a revision 6822-enabled Dahdi-Tools (see https://issues.asterisk.org/view.php?id=13897) with a Junghanns QuadBRI. This patch has now been merged into the trunk of DAHDI. 1. Do I still need qozap driver ? If positive, how is it recommended to get it ? No. 2. Which line should be included in /etc/dahdi/modules to have the appropriate driver loaded ? wcb4xxp . And if dahdi_hardware does not suggest that, it's a bug. My config is (with a Junghanns PCIe QuadBRI) : # asterisk -rx dahdi show version DAHDI Version: 2.2.0.2 Echo Canceller: # asterisk -rx core show version Asterisk 1.6.2.0-rc6 built by root @ foo on a i686 running Linux on 2009-11-25 00:04:47 UTC Dahdi tools version is revision 6822 (the one adding HFC cards support). # dahdi_hardware pci::06:04.0 qozap- 1397:08b4 Generic Cologne ISDN card I also tried with Dahdi tools revision 7664 (latest ?) and I still got the same qozap answer with dahdi_hardware. Are PCIe cards supported ? Is this a bug ? Cheers 3. The process I'm planning to use is : A- Hand edit /etc/dadhi/modules, /etc/dadhi/genconf_parameters and /etc/asterisk/chan_dadhi.conf. If you don't use dahdi_genconf, no point in editing genconf_parameters . In the trunk version, you won't need to edit it in order for it to provide proper version. B- Use dahdi_genconf to generate /etc/dadhi/system.conf and /etc/asterisk/dadhi_channels.conf. You forgot 'dahdi_genconf modules' :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 ___ -- 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] spandsp version
Hi! What version of spandsp is recommended to use when u compile asterisk-trunk? Best regards MAGNUS BENNGRD ___ -- 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] Fax throughput - Asterisk 1.6.1.9
Cyprus VoIP wrote: So, I enabled the full logger, and the strange thing I see is this message: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session It seems that this might be the reason Asterisk initiates a reINVITE with voice codecs, after connecting the 2 parties. Sorry, that's not the issue. That just means that chan_sip didn't destroy the internal RTP structures used for the audio part of the call when the call switched to T.38, which is only an optimization so we don't have to recreate them if the call switches back. -- 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
[asterisk-users] Dahdi_genconf does not generate NT/TE configuration
Hi, I'm using revision 6822 of Dahdi Tools. # dahdi_hardware pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P # asterisk -rx dahdi show version DAHDI Version: 2.2.0.2 Echo Canceller: OSLEC # cat /etc/dahdi/genconf_parameters ... pri_termtype SPAN/1 TE SPAN/2 TE SPAN/3 NT SPAN/4 NT # dahdi_genconf system # dahdi_genconf # cat /etc/dahdi/system.conf ... # Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS span=2,2,0,ccs,ami # termtype: te bchan=4-5 hardhdlc=6 echocanceller=oslec,4-5 # Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/CCS span=3,3,0,ccs,ami # termtype: te bchan=7-8 hardhdlc=9 echocanceller=oslec,7-8 # cat /etc/asterisk/dahdi-channels.conf ... ; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS group=1,12 context=remote switchtype = euroisdn signalling = bri_cpe channel = 4-5 context = default group = 63 ; Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/CCS group=1,13 context=remote switchtype = euroisdn signalling = bri_cpe channel = 7-8 context = default group = 63 ... As you can see, SPAN/3 is not configured for NT service, either in system.conf or dahdi-channels.conf. Did I miss something ? (Fortunately, when I hand edit both files, it does work). Cheers ___ -- 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] Get Queue values from dialplan (Was: queue_variables() function)
Olivier schrieb:
2009/12/4 Olivier oza-4...@myamail.com
Has someone successfully used this QUEUE_VARIABLES() function (in
1.6.2-rc7) ?
A previous question about it remainded unanswered (
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466).
http://lists.digium.com/pipermail/asterisk-users/2009-February/227122.html
http://lists.digium.com/pipermail/asterisk-users/2009-February/227127.html
https://issues.asterisk.org/view.php?id=14506
How can can you get current queue's length (ie maxlen) or waiting call
number from dialplan ?
Set(err=${QUEUE_VARIABLES(techsupport)});
Verbose(1,maxlen: ${QUEUEMAX});
Verbose(1,waiting calls: ${QUEUECALLS});
Philipp Kempgen
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] spandsp version
On 12/04/2009 06:54 PM, Magnus Benngård wrote: Hi! What version of spandsp is recommended to use when u compile asterisk-trunk? The next one, or if that hasn't been released yet, the current one. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get Queue values from dialplan (Was: queue_variables() function)
2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
2009/12/4 Olivier oza-4...@myamail.com
Has someone successfully used this QUEUE_VARIABLES() function (in
1.6.2-rc7) ?
A previous question about it remainded unanswered (
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466).
http://lists.digium.com/pipermail/asterisk-users/2009-February/227122.html
http://lists.digium.com/pipermail/asterisk-users/2009-February/227127.html
https://issues.asterisk.org/view.php?id=14506
How can can you get current queue's length (ie maxlen) or waiting call
number from dialplan ?
Set(err=${QUEUE_VARIABLES(techsupport)});
Verbose(1,maxlen: ${QUEUEMAX});
Verbose(1,waiting calls: ${QUEUECALLS});
That's Interesting because:
When includiing in my dialplan the same lines as yours, QUEUEMAX value
remains empty (while err equals -1).
With CLI, queue show techsupport says something like :
techsupportl has 0 calls (max 3) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/109 (Not in use) has taken no calls yet
No Callers
I also tried with and without setinterfacevar=yes or setqueuevar=yes.
Did you try with 1.6.2 ?
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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[asterisk-users] Get back in dialplan with number-parsing
I'd like to put a phone in a special context, where a test is made on its
business hours, then if so, proceed to the normal context to do whatever it
does with outgoing and local calls.
I've tried, just to go from one context to the next:
[specialoutgoing]
exten = _X.,1,noop(This is a special content)
exten = _X.,n,gotoiftime(?forbidden,1)
exten = _X.,n,goto(outgoing,${CALLERID(dnid)},1)
I use _X. to match anything, but if the call is allowed, I want to jump back in
the [outgoing] context and restart parsing the dialled number.
exten = _X.,n,goto(outgoing,${CALLERID(dnid)},1)
works only id the dialled extension exists precicely in outgoing context, not
in included contexts, and does not to pattern matching.
I can't include [outgoing] in [specialoutgoing], because the number has already
been matched by _X.
I don't want to rewrite the whole dialplan in [specialgoing] or to put the test
into the existing contexts.
Leif
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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9
Cyprus VoIP wrote: So, I enabled the full logger, and the strange thing I see is this message: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session It seems that this might be the reason Asterisk initiates a reINVITE with voice codecs, after connecting the 2 parties. Sorry, that's not the issue. That just means that chan_sip didn't destroy the internal RTP structures used for the audio part of the call when the call switched to T.38, which is only an optimization so we don't have to recreate them if the call switches back. Hi Kevin, Thank you for your support. If it's not related, why does Asterisk send again INVITE messages to both parties? How can this be prevented? I don't see more debug data prior to the new INVITE. Thanks. ___ -- 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] spandsp version
On Fri, Dec 04, 2009 at 09:58:40PM +0800, Steve Underwood wrote: On 12/04/2009 06:54 PM, Magnus Benngård wrote: Hi! What version of spandsp is recommended to use when u compile asterisk-trunk? The next one, or if that hasn't been released yet, the current one. Specifically? -- 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] Dahdi_genconf does not generate NT/TE configuration
On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote: Hi, I'm using revision 6822 of Dahdi Tools. # dahdi_hardware pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P # asterisk -rx dahdi show version DAHDI Version: 2.2.0.2 Echo Canceller: OSLEC # cat /etc/dahdi/genconf_parameters ... pri_termtype SPAN/1 TE SPAN/2 TE SPAN/3 NT SPAN/4 NT With BRI cards dahdi_genconf assumes that it uses whatever the card is jumpered for. -- 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] Fax throughput - Asterisk 1.6.1.9
Cyprus VoIP wrote: If it's not related, why does Asterisk send again INVITE messages to both parties? How can this be prevented? I don't see more debug data prior to the new INVITE. It's probably because you are using 1.6.1.9; that release (and older) had a 'feature' that allowed automatic switching back to audio from T.38 if one of the endpoints sent an audio packet. It turns out that wasn't a good idea, and it's been removed... but in later versions. You'll have to update to the latest release to get that fixed. -- 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] Get Queue values from dialplan (Was: queue_variables() function)
Olivier schrieb:
2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
How can can you get current queue's length (ie maxlen) or waiting call
number from dialplan ?
Set(err=${QUEUE_VARIABLES(techsupport)});
Verbose(1,maxlen: ${QUEUEMAX});
Verbose(1,waiting calls: ${QUEUECALLS});
When includiing in my dialplan the same lines as yours, QUEUEMAX value
remains empty (while err equals -1).
With CLI, queue show techsupport says something like :
techsupportl has 0 calls (max 3) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/109 (Not in use) has taken no calls yet
No Callers
I also tried with and without setinterfacevar=yes or setqueuevar=yes.
Did you try with 1.6.2 ?
Can't remember. Maybe I tested this with 1.6.0 or 1.6.1.
Philipp Kempgen
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Re: [asterisk-users] Dahdi_genconf does not generate NT/TE configuration
2009/12/4 Tzafrir Cohen tzafrir.co...@xorcom.com On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote: Hi, I'm using revision 6822 of Dahdi Tools. # dahdi_hardware pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P # asterisk -rx dahdi show version DAHDI Version: 2.2.0.2 Echo Canceller: OSLEC # cat /etc/dahdi/genconf_parameters ... pri_termtype SPAN/1 TE SPAN/2 TE SPAN/3 NT SPAN/4 NT With BRI cards dahdi_genconf assumes that it uses whatever the card is jumpered for. OK but I'm quite certain I set each jumper this way : jumper 1 and 2 in TE mode, 3 and 4 in NT modes with termination). How can I further check jumpers are correctly read ? As it worked correctly after I hand edited system.conf and dahdi-channels.cont with appropriate values and I connected port 1 to port 4 (and port 2 to 3), I would say there might an issue in card jumper detection. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 ___ -- 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] Dahdi_genconf does not generate NT/TE configuration
On Fri, Dec 04, 2009 at 04:06:29PM +0100, Olivier wrote: 2009/12/4 Tzafrir Cohen tzafrir.co...@xorcom.com On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote: Hi, I'm using revision 6822 of Dahdi Tools. # dahdi_hardware pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P # asterisk -rx dahdi show version DAHDI Version: 2.2.0.2 Echo Canceller: OSLEC # cat /etc/dahdi/genconf_parameters ... pri_termtype SPAN/1 TE SPAN/2 TE SPAN/3 NT SPAN/4 NT With BRI cards dahdi_genconf assumes that it uses whatever the card is jumpered for. OK but I'm quite certain I set each jumper this way : jumper 1 and 2 in TE mode, 3 and 4 in NT modes with termination). How can I further check jumpers are correctly read ? As it worked correctly after I hand edited system.conf and dahdi-channels.cont with appropriate values and I connected port 1 to port 4 (and port 2 to 3), I would say there might an issue in card jumper detection. The qozap driver showed it in name of the driver (or was it the description)? Hmm... what about wcb4xxp? -- 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] Rsrvd state and off hook dahdi issue
Hello again, Adding more information: Core show channels: Channel Location State Application(Data) DAHDI/4-1s...@national_mobile:1 Rsrvd(None) DAHDI/1-1s...@national_mobile:1 Rsrvd(None) Dahdi show channels: Chan ExtensionContext Language MOH Interpret pseudodefault default 1national_mobile pt default 3national_mobile pt default 4national_mobile pt default Thanks in advance, Alex ___ -- 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] Fax throughput - Asterisk 1.6.1.9
It's probably because you are using 1.6.1.9; that release (and older) had a 'feature' that allowed automatic switching back to audio from T.38 if one of the endpoints sent an audio packet. It turns out that wasn't a good idea, and it's been removed... but in later versions. You'll have to update to the latest release to get that fixed. Will do. Thanks for the explanation. ___ -- 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] Today in 30 minutes: VoIP on Social Networks
VoIP Users Conference begins in about 30 minutes to discuss the use of VoIP on social networks like Facebook. If you have any interest in this (or maybe you customers do?) please join us IRC anytime: #vuc on Freenode SIP see http://vuc.me for all the URI and PSTN numbers Skype:vuc.me or skype:ld.vuc.me (for reduced bandwidth) See you there. /r ___ -- 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] IAX2 Port issue
Trying to configure IAX for use I think I have everything set right. But my IAX phone wont connect. When I run wireshark I'm seeing this Note if above screenshot from wireshark does not show here is a link for it: http://img402.imageshack.us/i/tempe.jpg/ I've tried a variety of setups in my IAX.conf (they all end up with the same issue, tried just bindaddr=0.0.0.0 with bindport=4569, tried as in the below example specifying the port for the address and using a different once incase of conflict with something else I am unaware of. [general] bindport=4569 ; bindport and bindaddr may be specified ; ; NOTE: bindport must be specified BEFORE ; bindaddr or may be specified on a specific ; bindaddr if followed by colon and port ; (e.g. bindaddr=192.168.0.1:4569) bindaddr=192.168.17.140:4570 bindaddr=0.0.0.0; more than once to bind to multiple ; ; addresses, but the first will be the ; ; default ; The above being the most recent IAX.conf Below is what I get in the CLI whenever I reload for a change. egg*CLI iax2 reload == Parsing '/etc/asterisk/iax.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11087 set_config: Ignoring bindport on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload == Loaded firmware 'iaxy.bin' egg*CLI Any ideas? James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. image001.pngimage004.jpg___ -- 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] MWI count wrong when using IMAP and VM
[r...@voip ~]# asterisk -V Asterisk 1.6.1.11 When using the above version with IMAP VoiceMail integration when I leave a message my SNOM360 it shows 2 message waiting; yet when running voicemail show users from the Asterisk CLI it correctly reports 1. It would appear that when the VM is temporarily stored, and the VM is delivered by IMAP to the remote mail account, the MWI is being initiated with a incorrect count. I then delete the VM from either 1) the phone 2) the mail account the MWI goes blank and the message count shows 0 correctly. I am still trying to debug but any thoughts on this ? Here is how I have voicemail.conf :- [general] format=wav49 maxsecs=180 minsecs=5 skipms=3000 maxsilence=3 silencethreshold=128 maxlogins=3 imapserver=imap_server imapfolder=VoiceMail Office imapport=993 imapflags=ssl authuser=imap_user authpassword=imap_password [voicemail] 1001 = 1234,user,,,imapuser=u...@imap_server Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] DAHDI issues on 1.4.26.1
Hi, Running 1.4.26.1 here. I have installed TE420B card in my server, and followed the appropriate steps (as far as I know to configure it). This TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling type. When I dial out, I get this message: Dec 4 11:37:31] WARNING[27983]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I am using Dial(DHDI/g63/55|20) in my dialplan. Anyone who has a tip, I would appreciate. I seem to be stuck at the very basics, but since asterisk and other asterisk-related apps seem to see the card I don't get why I can't dial out. One thing that might be noted is that the default config files are old, so it might be Zaptel-specific. But I have gone through them and can't find what's missing or wrong. If Asterisk was dialing out but it didn't work, I'd assume a config problem, but it doesn't seem to recognize DAHDI as a channel type. Possibly relevant CLI output: CLI dahdi show status Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3OK 0 0 0 T4XXP (PCI) Card 0 Span 4RED0 0 0 (the 4th span is not live, so that seems like a good output). Another output: /etc/init.d/dahdi status ### Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF ClockSource 1 PRIClear (SWEC: MG2) 2 PRIClear (SWEC: MG2) 3 PRIClear (SWEC: MG2) 4 PRIClear (SWEC: MG2) 5 PRIClear (SWEC: MG2) 6 PRIClear (SWEC: MG2) 7 PRIClear (SWEC: MG2) 8 PRIClear (SWEC: MG2) 9 PRIClear (SWEC: MG2) 10 PRIClear (SWEC: MG2) 11 PRIClear (SWEC: MG2) 12 PRIClear (SWEC: MG2) 13 PRIClear (SWEC: MG2) 14 PRIClear (SWEC: MG2) 15 PRIClear (SWEC: MG2) 16 PRIClear (SWEC: MG2) 17 PRIClear (SWEC: MG2) 18 PRIClear (SWEC: MG2) 19 PRIClear (SWEC: MG2) 20 PRIClear (SWEC: MG2) 21 PRIClear (SWEC: MG2) 22 PRIClear (SWEC: MG2) 23 PRIClear (SWEC: MG2) 24 PRIHDLCFCS ### Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 B8ZS/ESF 25 PRIClear (SWEC: MG2) 26 PRIClear (SWEC: MG2) 27 PRIClear (SWEC: MG2) 28 PRIClear (SWEC: MG2) 29 PRIClear (SWEC: MG2) 30 PRIClear (SWEC: MG2) 31 PRIClear (SWEC: MG2) 32 PRIClear (SWEC: MG2) 33 PRIClear (SWEC: MG2) 34 PRIClear (SWEC: MG2) 35 PRIClear (SWEC: MG2) 36 PRIClear (SWEC: MG2) 37 PRIClear (SWEC: MG2) 38 PRIClear (SWEC: MG2) 39 PRIClear (SWEC: MG2) 40 PRIClear (SWEC: MG2) 41 PRIClear (SWEC: MG2) 42 PRIClear (SWEC: MG2) 43 PRIClear (SWEC: MG2) 44 PRIClear (SWEC: MG2) 45 PRIClear (SWEC: MG2) 46 PRIClear (SWEC: MG2) 47 PRIClear (SWEC: MG2) 48 PRIHDLCFCS ### Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 B8ZS/ESF 49 PRIClear (SWEC: MG2) 50 PRIClear (SWEC: MG2) 51 PRIClear (SWEC: MG2) 52 PRIClear (SWEC: MG2) 53 PRIClear (SWEC: MG2) 54 PRIClear (SWEC: MG2) 55 PRIClear (SWEC: MG2) 56 PRIClear (SWEC: MG2) 57 PRIClear (SWEC: MG2) 58 PRIClear (SWEC: MG2) 59 PRIClear (SWEC: MG2) 60 PRIClear (SWEC: MG2) 61 PRIClear (SWEC: MG2) 62 PRIClear (SWEC: MG2) 63 PRIClear (SWEC: MG2) 64 PRIClear (SWEC: MG2) 65 PRIClear (SWEC: MG2) 66 PRIClear (SWEC: MG2) 67 PRIClear (SWEC: MG2) 68 PRIClear (SWEC: MG2) 69 PRIClear (SWEC: MG2) 70 PRIClear (SWEC: MG2) 71 PRIClear (SWEC: MG2) 72 PRIHDLCFCS ### Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 B8ZS/ESF RED 73 PRIClear (SWEC: MG2) RED 74 PRIClear (SWEC: MG2) RED 75 PRIClear (SWEC: MG2) RED 76 PRIClear (SWEC: MG2) RED 77 PRIClear (SWEC: MG2) RED 78 PRIClear (SWEC: MG2) RED 79 PRIClear (SWEC: MG2) RED 80 PRIClear (SWEC: MG2) RED 81 PRIClear
Re: [asterisk-users] Dahdi_genconf does not generate NT/TE configuration
2009/12/4 Tzafrir Cohen tzafrir.co...@xorcom.com On Fri, Dec 04, 2009 at 04:06:29PM +0100, Olivier wrote: 2009/12/4 Tzafrir Cohen tzafrir.co...@xorcom.com On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote: Hi, I'm using revision 6822 of Dahdi Tools. # dahdi_hardware pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P # asterisk -rx dahdi show version DAHDI Version: 2.2.0.2 Echo Canceller: OSLEC # cat /etc/dahdi/genconf_parameters ... pri_termtype SPAN/1 TE SPAN/2 TE SPAN/3 NT SPAN/4 NT With BRI cards dahdi_genconf assumes that it uses whatever the card is jumpered for. OK but I'm quite certain I set each jumper this way : jumper 1 and 2 in TE mode, 3 and 4 in NT modes with termination). How can I further check jumpers are correctly read ? As it worked correctly after I hand edited system.conf and dahdi-channels.cont with appropriate values and I connected port 1 to port 4 (and port 2 to 3), I would say there might an issue in card jumper detection. The qozap driver showed it in name of the driver (or was it the description)? Hmm... what about wcb4xxp? I'm afraid I don't get it ... In this case, I'm using a single B410P card as shown with dahdi_hardware : # dahdi_hardware pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 ___ -- 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] DAHDI issues on 1.4.26.1
Forget it, found my issues. I have been looking for hours, but as soon as I write this I find it. dahdi-channels.conf wasn't included in chan_dahdi.conf. That being said, I have other issues now, but at least that one is fixed. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, December 04, 2009 11:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] DAHDI issues on 1.4.26.1 Hi, Running 1.4.26.1 here. I have installed TE420B card in my server, and followed the appropriate steps (as far as I know to configure it). This TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling type. When I dial out, I get this message: Dec 4 11:37:31] WARNING[27983]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I am using Dial(DHDI/g63/55|20) in my dialplan. Anyone who has a tip, I would appreciate. I seem to be stuck at the very basics, but since asterisk and other asterisk-related apps seem to see the card I don't get why I can't dial out. One thing that might be noted is that the default config files are old, so it might be Zaptel-specific. But I have gone through them and can't find what's missing or wrong. If Asterisk was dialing out but it didn't work, I'd assume a config problem, but it doesn't seem to recognize DAHDI as a channel type. Possibly relevant CLI output: CLI dahdi show status Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3OK 0 0 0 T4XXP (PCI) Card 0 Span 4RED0 0 0 (the 4th span is not live, so that seems like a good output). Another output: /etc/init.d/dahdi status ### Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF ClockSource 1 PRIClear (SWEC: MG2) 2 PRIClear (SWEC: MG2) 3 PRIClear (SWEC: MG2) 4 PRIClear (SWEC: MG2) 5 PRIClear (SWEC: MG2) 6 PRIClear (SWEC: MG2) 7 PRIClear (SWEC: MG2) 8 PRIClear (SWEC: MG2) 9 PRIClear (SWEC: MG2) 10 PRIClear (SWEC: MG2) 11 PRIClear (SWEC: MG2) 12 PRIClear (SWEC: MG2) 13 PRIClear (SWEC: MG2) 14 PRIClear (SWEC: MG2) 15 PRIClear (SWEC: MG2) 16 PRIClear (SWEC: MG2) 17 PRIClear (SWEC: MG2) 18 PRIClear (SWEC: MG2) 19 PRIClear (SWEC: MG2) 20 PRIClear (SWEC: MG2) 21 PRIClear (SWEC: MG2) 22 PRIClear (SWEC: MG2) 23 PRIClear (SWEC: MG2) 24 PRIHDLCFCS ### Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 B8ZS/ESF 25 PRIClear (SWEC: MG2) 26 PRIClear (SWEC: MG2) 27 PRIClear (SWEC: MG2) 28 PRIClear (SWEC: MG2) 29 PRIClear (SWEC: MG2) 30 PRIClear (SWEC: MG2) 31 PRIClear (SWEC: MG2) 32 PRIClear (SWEC: MG2) 33 PRIClear (SWEC: MG2) 34 PRIClear (SWEC: MG2) 35 PRIClear (SWEC: MG2) 36 PRIClear (SWEC: MG2) 37 PRIClear (SWEC: MG2) 38 PRIClear (SWEC: MG2) 39 PRIClear (SWEC: MG2) 40 PRIClear (SWEC: MG2) 41 PRIClear (SWEC: MG2) 42 PRIClear (SWEC: MG2) 43 PRIClear (SWEC: MG2) 44 PRIClear (SWEC: MG2) 45 PRIClear (SWEC: MG2) 46 PRIClear (SWEC: MG2) 47 PRIClear (SWEC: MG2) 48 PRIHDLCFCS ### Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 B8ZS/ESF 49 PRIClear (SWEC: MG2) 50 PRIClear (SWEC: MG2) 51 PRIClear (SWEC: MG2) 52 PRIClear (SWEC: MG2) 53 PRIClear (SWEC: MG2) 54 PRIClear (SWEC: MG2) 55 PRIClear (SWEC: MG2) 56 PRIClear (SWEC: MG2) 57 PRIClear (SWEC: MG2) 58 PRIClear (SWEC: MG2) 59 PRIClear (SWEC: MG2) 60 PRIClear (SWEC: MG2) 61 PRIClear (SWEC: MG2) 62 PRIClear (SWEC: MG2) 63 PRIClear (SWEC: MG2) 64 PRIClear (SWEC: MG2) 65 PRIClear (SWEC: MG2) 66 PRIClear (SWEC: MG2) 67 PRIClear (SWEC: MG2) 68 PRIClear (SWEC: MG2) 69 PRIClear (SWEC: MG2) 70 PRI
Re: [asterisk-users] IAX2 Port issue
On 4 Dec 2009, at 16:37, James A. Shigley wrote: egg*CLI iax2 reload == Parsing '/etc/asterisk/iax.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11087 set_config: Ignoring bindport on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload == Loaded firmware 'iaxy.bin' egg*CLI Its a notice rather than a warning. I doubt thats your problem. Wireshark doesn't tell us enough, What are your network addresses? You at least need to tell us which end is which, and your subnet etc. Firewall info good too. 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] Get Queue values from dialplan (Was: queue_variables() function)
2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
How can can you get current queue's length (ie maxlen) or waiting call
number from dialplan ?
Set(err=${QUEUE_VARIABLES(techsupport)});
Verbose(1,maxlen: ${QUEUEMAX});
Verbose(1,waiting calls: ${QUEUECALLS});
When includiing in my dialplan the same lines as yours, QUEUEMAX value
remains empty (while err equals -1).
With CLI, queue show techsupport says something like :
techsupportl has 0 calls (max 3) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/109 (Not in use) has taken no calls yet
No Callers
I also tried with and without setinterfacevar=yes or setqueuevar=yes.
Did you try with 1.6.2 ?
Can't remember. Maybe I tested this with 1.6.0 or 1.6.1.
OK !
I'll also try with one of these and see if things behave the same ...
Philipp Kempgen
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Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] IAX2 Port issue
192.168.16.3 is my desk 17.140 is * 192.168.16.0/21 is the subnet (255.255.248.0) Firewall isn't an issue here, that I can see for sure. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Friday, December 04, 2009 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX2 Port issue On 4 Dec 2009, at 16:37, James A. Shigley wrote: egg*CLI iax2 reload == Parsing '/etc/asterisk/iax.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11087 set_config: Ignoring bindport on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload == Loaded firmware 'iaxy.bin' egg*CLI Its a notice rather than a warning. I doubt thats your problem. Wireshark doesn't tell us enough, What are your network addresses? You at least need to tell us which end is which, and your subnet etc. Firewall info good too. 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 ___ -- 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] IAX2 Port issue
Ok, check if it is actually listening using netstat? Steve On 4 Dec 2009, at 17:17, James A. Shigley wrote: 192.168.16.3 is my desk 17.140 is * 192.168.16.0/21 is the subnet (255.255.248.0) Firewall isn't an issue here, that I can see for sure. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com ] On Behalf Of Steve Howes Sent: Friday, December 04, 2009 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX2 Port issue On 4 Dec 2009, at 16:37, James A. Shigley wrote: egg*CLI iax2 reload == Parsing '/etc/asterisk/iax.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11087 set_config: Ignoring bindport on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload [Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring bindaddr on reload == Loaded firmware 'iaxy.bin' egg*CLI Its a notice rather than a warning. I doubt thats your problem. Wireshark doesn't tell us enough, What are your network addresses? You at least need to tell us which end is which, and your subnet etc. Firewall info good too. 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 ___ -- 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] MWI count wrong when using IMAP and VM
Following up on this if I leave a second message then the WMI count goes to 4. When I check the voicemail directory on the server I see :- [r...@voip 1001]# ls -lR .: total 20 drwxr-xr-x 2 root root 4096 Dec 4 17:49 INBOX drwxr-xr-x 2 root root 4096 Oct 8 21:02 Old drwxr-xr-x 2 root root 4096 May 13 2009 temp drwxr-xr-x 2 root root 4096 Dec 4 17:49 tmp drwxr-xr-x 2 root root 4096 Dec 4 15:24 VoiceMail Office ./INBOX: total 0 ./Old: total 0 ./temp: total 0 ./tmp: total 0 ./VoiceMail Office: total 0 but from the CLI I get :- voip*CLI voicemail show users ContextMbox User Zone NewMsg voicemail 1001 user2 Best Regards, - --[ UxBoD ]-- ux...@splatnix.net wrote: | [r...@voip ~]# asterisk -V | Asterisk 1.6.1.11 | | When using the above version with IMAP VoiceMail integration when I | leave a message my SNOM360 it shows 2 message waiting; yet when | running voicemail show users from the Asterisk CLI it correctly | reports 1. | | It would appear that when the VM is temporarily stored, and the VM is | delivered by IMAP to the remote mail account, the MWI is being | initiated with a incorrect count. | | I then delete the VM from either 1) the phone 2) the mail account the | MWI goes blank and the message count shows 0 correctly. | | I am still trying to debug but any thoughts on this ? | | Here is how I have voicemail.conf :- | | [general] | format=wav49 | maxsecs=180 | minsecs=5 | skipms=3000 | maxsilence=3 | silencethreshold=128 | maxlogins=3 | imapserver=imap_server | imapfolder=VoiceMail Office | imapport=993 | imapflags=ssl | authuser=imap_user | authpassword=imap_password | | [voicemail] | 1001 = 1234,user,,,imapuser=u...@imap_server | | Best Regards, | | | -- | This message has been scanned for viruses and | dangerous content and is believed to be clean. | | SplatNIX IT Services :: Innovation through collaboration | | | ___ | -- Bandwidth and Colocation Provided by http://www.api-digital.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] MWI count wrong when using IMAP and VM
as soon as I delete the two messages I receive in the console :- [Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP Warning: Unknown message data: 1 EXPUNGE [Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP Warning: Unknown message data: 1 EXPUNGE Best Regards, - --[ UxBoD ]-- ux...@splatnix.net wrote: | [r...@voip ~]# asterisk -V | Asterisk 1.6.1.11 | | When using the above version with IMAP VoiceMail integration when I | leave a message my SNOM360 it shows 2 message waiting; yet when | running voicemail show users from the Asterisk CLI it correctly | reports 1. | | It would appear that when the VM is temporarily stored, and the VM is | delivered by IMAP to the remote mail account, the MWI is being | initiated with a incorrect count. | | I then delete the VM from either 1) the phone 2) the mail account the | MWI goes blank and the message count shows 0 correctly. | | I am still trying to debug but any thoughts on this ? | | Here is how I have voicemail.conf :- | | [general] | format=wav49 | maxsecs=180 | minsecs=5 | skipms=3000 | maxsilence=3 | silencethreshold=128 | maxlogins=3 | imapserver=imap_server | imapfolder=VoiceMail Office | imapport=993 | imapflags=ssl | authuser=imap_user | authpassword=imap_password | | [voicemail] | 1001 = 1234,user,,,imapuser=u...@imap_server | | Best Regards, | | | -- | This message has been scanned for viruses and | dangerous content and is believed to be clean. | | SplatNIX IT Services :: Innovation through collaboration | | | ___ | -- Bandwidth and Colocation Provided by http://www.api-digital.com -- | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] spandsp version
magnus, simple answer: just use the latest version available. and if something is not working inside the t.30/t.38 protocol, try the latest spanpshot: http://www.soft-switch.org/downloads/snapshots/spandsp/?C=M;O=Dand if something i still not working, give a good description how to reproduce the problem. Kristijan ___ -- 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] No audio - using g729 codec altogether
Hi, I am facing terrible issue regarding no audio/voice on both sides. I am using g729 codec on two machines and carrier also supports g729 codec. I can see the RTP traffic flowing but there is no audio. Call is going from Server 1 to Server 2. I can see the established SIP channels on Server but still no audio C.L.M.37 = Global Address **-Server-1* X.Y.X.55 = LAN Address of **-Server-1* M.G.W.23 = Media Gateway of Carrier A.B.C.136 = Global Address **-Server-2* **-Server-1* codec and format used: codec_g729.so format_g729.so 126.475451X.Y.X.55 - M.G.W.23RTP Payload type=ITU-T G.729, SSRC=1224682667, Seq=171, Time=171288 126.495804 A.B.C.136 - C.L.M.37 RTP Payload type=ITU-T G.729, SSRC=1406269818, Seq=17753, Time=171608 126.495833X.Y.X.55 - M.G.W.23RTP Payload type=ITU-T G.729, SSRC=1224682667, Seq=172, Time=171448 126.515405 A.B.C.136 - C.L.M.37 RTP Payload type=ITU-T G.729, SSRC=1406269818, Seq=17754, Time=171768 126.515435X.Y.X.55 - M.G.W.23RTP Payload type=ITU-T G.729, SSRC=1224682667, Seq=173, Time=171608 126.535204 A.B.C.136 - C.L.M.37 RTP Payload type=ITU-T G.729, SSRC=1406269818, Seq=17755, Time=171928 126.535423X.Y.X.55 - M.G.W.23RTP Payload type=ITU-T G.729, SSRC=1224682667, Seq=174, Time=171768 126.555461 A.B.C.136 - C.L.M.37 RTP Payload type=ITU-T G.729, SSRC=1406269818, Seq=17756, Time=172088 126.79X.Y.X.55 - M.G.W.23RTP Payload type=ITU-T G.729, SSRC=1224682667, Seq=175, Time=171928 **-Server-2* Codec used : codec_g729-ast12-gcc4-glibc-x86_64-pentium4.so 101.374796 A.B.C.136 - C.L.M.37 RTP PT=ITU-T G.729, SSRC=0x53D1F97A, Seq=17785, Time=176728 101.389644 213.166.5.134 - A.B.C.136 RTP PT=ITU-T G.729, SSRC=0x15130586, Seq=55472, Time=879827237 101.389665 A.B.C.136 - C.L.M.37 RTP PT=ITU-T G.729, SSRC=0x53D1F97A, Seq=17786, Time=176888 101.409653 213.166.5.134 - A.B.C.136 RTP PT=ITU-T G.729, SSRC=0x15130586, Seq=55473, Time=879827397 101.409674 A.B.C.136 - C.L.M.37 RTP PT=ITU-T G.729, SSRC=0x53D1F97A, Seq=17787, Time=177048 101.429709 213.166.5.134 - A.B.C.136 RTP PT=ITU-T G.729, SSRC=0x15130586, Seq=55474, Time=879827557 101.429723 A.B.C.136 - C.L.M.37 RTP PT=ITU-T G.729, SSRC=0x53D1F97A, Seq=17788, Time=177208 101.454956 213.166.5.134 - A.B.C.136 RTP PT=ITU-T G.729, SSRC=0x15130586, Seq=55475, Time=879827717 Any one has any idea why it is behaving so. /ag ___ -- 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] DAHDI outgoing
Hi, I'm having alot of trouble understanding how to use dialplans for outgoing calls on Dahdi. Context : I have 3 TI spans, so 69 voice channels and three D channels (24,48,72). This is on a TE420B from Digium, if it matters. Here are my (apparently simple) questions in no particular order: 1) Dial(DAHDI/55|20) doesn't work. But Dial(DAHDI/42/55|20) does work. How is this 42 parameter used? I see plenty of examples around, but no explanations. Is this the channel? If so, why doesn't DAHDI/3 work? 2) dahdi show channels in the CLI show my inbound used channels correctly, but not my outbound. My outbound never show up, even during a conversation. Thanks for helping me figure this out. Mike ___ -- 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] DAHDI - Split data voice use
Can any Digium E1 cards be used for split data/voice use? ___ -- 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] DAHDI outgoing
Simple explanation for #1; it's dial tech/port/number. Dahdi/3 would open DAHDI port 3 for an outgoing call. For #2, you should be using core show channels instead of dahdi show channels. Dsc shows the lines that are available to asterisk, csc shows the ones in use. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, December 04, 2009 1:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] DAHDI outgoing Hi, I'm having alot of trouble understanding how to use dialplans for outgoing calls on Dahdi. Context : I have 3 TI spans, so 69 voice channels and three D channels (24,48,72). This is on a TE420B from Digium, if it matters. Here are my (apparently simple) questions in no particular order: 1) Dial(DAHDI/55|20) doesn't work. But Dial(DAHDI/42/55|20) does work. How is this 42 parameter used? I see plenty of examples around, but no explanations. Is this the channel? If so, why doesn't DAHDI/3 work? 2) dahdi show channels in the CLI show my inbound used channels correctly, but not my outbound. My outbound never show up, even during a conversation. Thanks for helping me figure this out. Mike ___ -- 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] DAHDI outgoing
Thanks a lot. That helped. As for #2, dahdi show channels still lists channel 71 (in my particular case) even though it is in use (core show channels shows it being used). It's just that the extension is empty in dahdi show channels. i.e.: Chan Extension Context Language MOH Interpret pseudodefaultdefault 1inbounddefault 2inbounddefault 3inbounddefault 4inbounddefault 5inbounddefault 6inbounddefault 7inbounddefault 8inbounddefault 9inbounddefault 10inbounddefault 11inbounddefault 12inbounddefault 13inbounddefault 14inbounddefault 15inbounddefault 16inbounddefault 17inbounddefault 18inbounddefault 19inbounddefault 20inbounddefault 21inbounddefault 22inbounddefault 23inbounddefault 25inbounddefault 26inbounddefault 27inbounddefault 28inbounddefault 29inbounddefault 30inbounddefault 31inbounddefault 32inbounddefault 33inbounddefault 34inbounddefault 35inbounddefault 36inbounddefault 37inbounddefault 38inbounddefault 39inbounddefault 40inbounddefault 41inbounddefault 42inbounddefault 43inbounddefault 44inbounddefault 45inbounddefault 46inbounddefault 47inbounddefault 49inbounddefault 50inbounddefault 51inbounddefault 52inbounddefault 53inbounddefault 54inbounddefault 55inbounddefault 56inbounddefault 57inbounddefault 58inbounddefault 59inbounddefault 60inbounddefault 61inbounddefault 62inbounddefault 63inbounddefault 64inbounddefault 65inbounddefault 66inbounddefault 67inbounddefault 68inbounddefault 69inbounddefault 70inbounddefault 71inbounddefault If it wouldn't show in dahdi show channels, I'd understand your explanation. But it does, so is it in use or not? Or am I misunderstanding the displayed information? What I was looking for was a list of channels that shows which one is currently busy, and ideally also a way to know how many channels are used per span (although I could work it out from the former if it was available) Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, December 04, 2009 14:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] DAHDI outgoing Simple explanation for #1; it's dial tech/port/number. Dahdi/3 would open DAHDI port 3 for an outgoing call. For #2, you should be using
Re: [asterisk-users] DAHDI outgoing
On my * 1.6.0.13 box I see this: dahdi show channels Chan Extension Context Language MOH InterpretBlocked State pseudononesaiden default In Service 1 415111 from-outsideen default In Service 2from-outsideen default In Service 3from-outsideen default In Service 4from-outsideen default In Service 5from-outsideen default In Service 6from-outsideen default In Service 7from-outsideen default In Service 8from-outsideen default In Service 9from-outsideen default In Service 10from-outsideen default In Service 11from-outsideen default In Service 12from-outsideen default In Service 13from-outsideen default In Service 14from-outsideen default In Service 15from-outsideen default In Service 16from-outsideen default In Service 17from-outsideen default In Service 18from-outsideen default In Service 19from-outsideen default In Service 20from-outsideen default In Service 21from-outsideen default In Service 22from-outsideen default In Service 23from-outsideen default In Service In this case I had an inbound call to one of my DID numbers for this PRI line. It showed the DID on the port 1 line. If I made an outbound call nothing showed for the port used. To see that I needed to do the command core show channels and then I saw this for the outbound channel in use: DAHDI/23-1 (None) Up AppDial((Outgoing Line)) I do not see a way to do any summary reports directly in CLI. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 4, 2009, at 11:51 AM, Mike wrote: Or am I misunderstanding the displayed information? What I was looking for was a list of channels that shows which one is currently busy, and ideally also a way to know how many channels are used per span (although I could work it out from the former if it was 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
Re: [asterisk-users] The SIP in the Mobile Phones are not able to register on asterisk
Dear Xavier; Actually I beleive you put me in the right channel, but for me realm is something new to be used. I did not try it at all before. I read some about it, but still I am not familiar with it If you can help me in the realm, I will appreciate this: 1) What is the relation between the username in the realm and the username I configure it in the sip.conf? 2) Where I configure the realm value? Is it in the sip.conf or other? 3) When actually we use the realm? Your kindly help is high appreciated. Regards Bilal - Have you set the realm in the sip settings in the mobile? Default one is asterisk . It's important too, defining Registration to Always on, because if not, it doesn't enable the wifi connection. Finally, don't enable compression and security ___ -- 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] Audio issue in skype for asterisk
we have a similar problem. When we try to make two skype-calls at a time, only one of them has working audio. For this to happen, both calls must be ringing at the same time. Does anyone know how to fix this? I have fixed this issue and it will be in the 1.0.7 release which is currently in PQ for testing (gotta make sure I didn't introduce any new bugs). I would guess that it will probably be available next week sometime. Terry ___ -- 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] DAHDI outgoing
Thank you, at least I am getting the same thing. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Friday, December 04, 2009 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI outgoing On my * 1.6.0.13 box I see this: dahdi show channels Chan Extension Context Language MOH InterpretBlocked State pseudononesaiden default In Service 1 415111 from-outsideen default In Service 2from-outsideen default In Service 3from-outsideen default In Service 4from-outsideen default In Service 5from-outsideen default In Service 6from-outsideen default In Service 7from-outsideen default In Service 8from-outsideen default In Service 9from-outsideen default In Service 10from-outsideen default In Service 11from-outsideen default In Service 12from-outsideen default In Service 13from-outsideen default In Service 14from-outsideen default In Service 15from-outsideen default In Service 16from-outsideen default In Service 17from-outsideen default In Service 18from-outsideen default In Service 19from-outsideen default In Service 20from-outsideen default In Service 21from-outsideen default In Service 22from-outsideen default In Service 23from-outsideen default In Service In this case I had an inbound call to one of my DID numbers for this PRI line. It showed the DID on the port 1 line. If I made an outbound call nothing showed for the port used. To see that I needed to do the command core show channels and then I saw this for the outbound channel in use: DAHDI/23-1 (None) Up AppDial((Outgoing Line)) I do not see a way to do any summary reports directly in CLI. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 4, 2009, at 11:51 AM, Mike wrote: Or am I misunderstanding the displayed information? What I was looking for was a list of channels that shows which one is currently busy, and ideally also a way to know how many channels are used per span (although I could work it out from the former if it was 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] How to use SIP hints and BLF for realtime extensions on Aastra phones?
Hi, I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. Is there any work around? On voip-info I read that Snom phones can use BLF without using hints. Is it possible to do similar on Aastra phones? Any guidance will be highly appreciated. -- Zeeshan A Zakaria ___ -- 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
