Re: [asterisk-users] Does Playback will answer the call?
Zhang Shukun wrote: hi, all in my test,it shows Playback will answer the call automaticly, but i don't want to so. i will use answer function to answer the call. could you help me ? core show application Playback Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE410P Spans offline/red after power down/restart
I've just encountered an odd problem with our Digium TE410P card and was wondering if anyone has experienced something similar before. We utilize all 4 ports with 2 of them connected to the PSTN as E1 with the second 2 ports connecting to a device which accepts T1. We are essentially acting as an E1 to T1 converter. This machine has been operational for close to 6 months without a problem. Today after restarting the box after a scheduled power outage none of the spans on the TE410P were coming up. All were reporting RED alarms. Nothing had changed on the server and nothing had been changed at the telco side. We quickly removed the card and checked the jumpers, suspecting that it may somehow have 'changed' it's state. Checking the card showed that all ports were open, which is what we expected. Re-setting the card done nothing to resolve our problem. We tested the line from the telco by connecting to our PBX, which can accept a T1 connection. This worked fine, ruling out any telco based problems. lspci showed that the card was being detected and 'zap show status' showed all the spans and channels configured as they should have been. Testing the ports and observing how things were behaving with our PBX it appeared that the TE410P ports were able to send out a signal, but for some reason, it could not accept a signal. After some trial and error, and more than a few reboots we finally found a solution of sorts. Here is what we had to do: * Power down the machine * Disconnect all cables from the ports * Power up * After everything has started up again re-connect the ports This resulted in all the ports going green and becoming operational. Has anyone every experienced this or something similar before? Everything is working fine now, but I would rather not have to visit our data centers everytime we have to power cycle one of our machines. Could this be a config issue in the way the zaptel driver is loaded? Here are the versions we are running: Asterisk 1.4.18.1 Zaptel 1.4.9.2-48 Libpri 1.4.3-19 This is all bundled in the SME Server SARK distrubution. I can provide a copy of our zapata/zaptel configs if necessary. Any ideas/pointers/anecdotes welcomed, Cheers, Conor -- _ -- 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] TE410P Spans offline/red after power down/restart
Le 22/02/2010 09:28, Conor McTernan a écrit : I've just encountered an odd problem with our Digium TE410P card and was wondering if anyone has experienced something similar before. There is one similar request on this list from a few weeks back iirc We quickly removed the card and checked the jumpers, suspecting that it may somehow have 'changed' it's state. Checking the card showed that all ports were open, which is what we expected. Re-setting the card done nothing to resolve our problem. You did not disconnect cables by doing so ? How much time did you let the card disconnected from everything, at least 30s ? * Power down the machine * Disconnect all cables from the ports * Power up * After everything has started up again re-connect the ports If i'm not misleading isdn lines are powered, not huge amount of power but still more than an ethernet cable. Maybe the power was sufficient to held the card in a misconfigured state Here are the versions we are running: Asterisk 1.4.18.1 Zaptel 1.4.9.2-48 Libpri 1.4.3-19 Or you are simply using very outdated drivers/software :) -- _ -- 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] Help with Dictate app
I find there are only few mails other than mine, which hasn't been replied to. Have I put the question in the wrong mailing list ? Sorry for being impatient. -- Jayesh Jayan The box said Requires Windows 95, NT, or better, so I installed Linux. Visit my homepage @ http://www.jayeshjayan.com On Wed, Feb 17, 2010 at 8:27 PM, Jayesh Jayan jayesh.ja...@gmail.comwrote: Hello One and All, I am a Linux admin, new to asterisk. I have been assigned the task of setting up a dictation server for the company I work for. Our company is into transcription. Currently we are using dictation server, which is provided by another company. Now we have decided to have our own dictation server. I have installed asterisk and have gone through many documentation and guidance's available online and was able to create a dictation server. But it doesn't meet all the requirements which we need. I have used the dictate app for the dictation purpose. The dictate application have a predefined set of functions for each of the phone keypads. I wish to change those, is it possible ? Also there is a need for the keypad settings to changed dynamically for clients logging in (authenticated by a agi script and according to client preference). Would it be possible as well ? Any help/guidance in this regard will be greatly helpful. -- Jayesh Jayan The box said Requires Windows 95, NT, or better, so I installed Linux. Visit my homepage @ http://www.jayeshjayan.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax, T38 and NAT
Yes, when I added t38pt_usertpsource=yes to the NAT'ed fax everything works! Big thanks Johann! On Sun, 21 Feb 2010 17:22:40 +0100, Magnus Benngård wrote: t38pt_usertpsource=yes seems to do the trick, switches to T38 and fax seems to go through (cant be 100% sure, the fax i am sending to is 500 km avay from me, but i dont get any errors and my fax thinks everything is ok, so I cross my fingers),,, On Sun, 21 Feb 2010 16:36:42 +0100, Johann Steinwendtner wrote: Magnus Benngård wrote: Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are wrong. UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPTL (SIP/0851711201): packet from 10.242.20.149:16434 (type 0, seq 0, len 6) 90.230.92.67 is WAN ip of 0197673581's router. 10.242.20.149 is ip of 0851711201's ATA (SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz callerid=Input Interior Stockholm (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=yes context=inputinterior.se directmedia=yes dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=no qualify=yes sendrpid=pai t38pt_udptl=yes,fec transport=udp trustrpid=yes type=friend videosupport=no [0851711290] secret=xyz callerid=Input Interior Sundbyberg (fax) ... rest is the same as [0851711201] Regards, Magnus Maybe you should give t38pt_usertpsource=yes a try. Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 to remote AGI server
22 feb 2010 kl. 07.23 skrev Tilghman Lesher:
open audio {tcp|udp} hostname portno
close audio
If you design something now, I would strongly suggest that we stop using
audio as an attribute. Each call will have multiple media streams - and
already have. You need to be able to select which one, and possibly open
multiple streams - audio, video, fax, text. In the future, we'll hopefully have
the ability to run multiple of each category, so I would not design this
feature for a single audio stream to be open for future use.
Just my 10 öre. :-)
/O
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Re: [asterisk-users] HFC-S card
Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if that helps. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] HFC-S card
On 22 February 2010 10:26, Per Jessen p...@computer.org wrote: Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if that helps. /Per Jessen, Zürich Not meaning to hi-jack this thread, I’m not getting a response to my “mISDN (HFC-S) and TDM400P - mISDN: ISAC XDU no TX_BUSY” thread so I thought I would check my understanding here. Am I correct in saying that if you can use a standard HFC-S BRI card with DAHDi 2.2.1, which means you no longer require mISDN? -- _ -- 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 back the BYE code gotten on second leg
I have a business problem that is killing me. I do SIP2SIP, only. I place a call after receiving the incoming request, and I need to send a Hangup(Code) to the caller, based on the result of the outbound leg. How can I do that in Asterisk? Is that even possible at all? I can use Hangup(code), but how do I extract it from the received BYE? This is only for calls that fail to connect on the outbound leg. Philip -- _ -- 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] Sending back the BYE code gotten on second leg
On 22 Feb 2010, at 11:16, CDR wrote: I have a business problem that is killing me. I do SIP2SIP, only. I place a call after receiving the incoming request, and I need to send a Hangup(Code) to the caller, based on the result of the outbound leg. How can I do that in Asterisk? Is that even possible at all? I can use Hangup(code), but how do I extract it from the received BYE? This is only for calls that fail to connect on the outbound leg. Just do Hangup(), I think it should pass it automatically as long as you never did Answer()... Might be wrong. S -- _ -- 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] AMI Originate differences between 1.4 and 1.6.1
Folks, I am strugging with Asterisk 1.4 Vs 1.6 differences over AMI Originate? Here is the pastebin... http://pastebin.ca/1805594 Not sure why the local channel won't send to context while the remote channel does. Worked fine in 1.4 but 1.6.1 has issues. Any help? Ritesh -- _ -- 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] Denying call transfer to certain extensions
Hi all, Is there a way to deny call transfers to certain extensions? Thanks, Ahmed Ossama -- _ -- 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] HFC-S card
On 2/22/2010 10:26 AM, Per Jessen wrote: Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if that helps. /Per Jessen, Zürich I have use this howto http://www.voip-info.org/wiki/view/Asterisk+zaphfc; , but i can´t put the card working in nt ptp mode. Can you explain me how i have to do that? Do you have any howto to make the card work in nt ptp mode? thanks for answer /Pedro Santos -- _ -- 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] HFC-S card
On 2/22/2010 7:36 AM, Tzafrir Cohen wrote: On Sun, Feb 21, 2010 at 07:55:39PM +, Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? Which version of Asterisk do you use? Which channel driver? I have use this howto http://www.voip-info.org/wiki/view/Asterisk+zaphfc; Pedro Santos -- _ -- 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] Help with Dictate app
Jayesh Jayan wrote: I find there are only few mails other than mine, which hasn't been replied to. Have I put the question in the wrong mailing list ? Not the wrong mailing list, but most likely nobody has any answers for you. I personally have never used the application. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with Dictate app
Doug, Thank you for your update. Google results also reveal very less number of users for this app. By the if we assume it is some other app, and we have to change the default keypad settings, how do we go about changing it ? Do we have to alert the code ? -- Jayesh Jayan The box said Requires Windows 95, NT, or better, so I installed Linux. Visit my homepage @ http://www.jayeshjayan.com On Mon, Feb 22, 2010 at 6:19 PM, Doug Lytle supp...@drdos.info wrote: Jayesh Jayan wrote: I find there are only few mails other than mine, which hasn't been replied to. Have I put the question in the wrong mailing list ? Not the wrong mailing list, but most likely nobody has any answers for you. I personally have never used the application. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] HFC-S card
On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote: On 2/22/2010 10:26 AM, Per Jessen wrote: Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if that helps. /Per Jessen, Zürich I have use this howto http://www.voip-info.org/wiki/view/Asterisk+zaphfc; , but i can´t put the card working in nt ptp mode. Can you explain me how i have to do that? Do you have any howto to make the card work in nt ptp mode? thanks for answer Short answert: signalling = bri_net Longer answer: That page is outdated (hmm, and I didn't get to update it :-( ) Nowadays (as of Asterisk 1.6.0) BRI support is included in Asterisk. The zaphfc driver, though, is still not included in DAHDI. It's maintained, though. The version included in the Debian packages is taken from http://git.tzafrir.org.il/?p=dahdi-extra.git;a=summary . Either way (bristuff or Asterisk = 1.6.0) to use BRI PTP NT in chan_dahdi you should set: signalling = bri_net for the span's channels in /etc/asterisk/chan_dahdi.conf . -- 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
[asterisk-users] 4 PCIe cards in one asterisk server
Hi, Does anybody have any experience with asterisk where are four PCIe cards are used in one server (TE420). So you can have max 4 * 4 * 30 channels = 480 channels used. Regards, Arjan Kroon Mobillion BV -- _ -- 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] Help with Dictate app
Jayesh Jayan wrote: By the if we assume it is some other app, and we have to change the default keypad settings, how do we go about changing it ? Do we have to alert the code ? I'm guessing that you'd have to modify the code. Not knowing this particular application and not having any programming experience, this is all speculation. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S card
On 22 February 2010 13:02, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Nowadays (as of Asterisk 1.6.0) BRI support is included in Asterisk. The zaphfc driver, though, is still not included in DAHDI. It's maintained, though. The version included in the Debian packages is taken from http://git.tzafrir.org.il/?p=dahdi-extra.git;a=summary . I'm using CentOS5.4, can anyone advise how I can make DAHDi work with a generic HFC-S card? -- _ -- 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] Help with Dictate app
Thank you, Doug. Let me try in that direction. -- Jayesh Jayan The box said Requires Windows 95, NT, or better, so I installed Linux. Visit my homepage @ http://www.jayeshjayan.com On Mon, Feb 22, 2010 at 6:48 PM, Doug Lytle supp...@drdos.info wrote: Jayesh Jayan wrote: By the if we assume it is some other app, and we have to change the default keypad settings, how do we go about changing it ? Do we have to alert the code ? I'm guessing that you'd have to modify the code. Not knowing this particular application and not having any programming experience, this is all speculation. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Denying call transfer to certain extensions
Follow-me will most likely be your best bet for this trick. Say you have extensions 100, 101 and 102. 100 is the receptionist, 101 is sales and 102 is the boss, who doesn't want to be disturbed. If you set up followme on 102 to go to voicemail or whatever, 102 won't ring. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Ossama Sent: Monday, February 22, 2010 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Denying call transfer to certain extensions Hi all, Is there a way to deny call transfers to certain extensions? Thanks, Ahmed Ossama -- _ -- 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] Unrecognized prilocaldialplan NPI modifier
On Wed, Feb 17, 2010 at 6:53 PM, Tilghman Lesher tles...@digium.com wrote: Oh, right, priLOCALdialplan. What's in CALLERID(num) ? Legitimate characters for the PSTN are numbers (and ABCD) only, so other characters are invalid, making them candidates for usage in modifying prilocaldialplan. Hi, Thanks for the help! It's was because callerid(num) was set to the string 'unknown' in a agi script, in some cases where the user calling in didn't have a callerid. Håkon -- _ -- 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] 4 PCIe cards in one asterisk server
Not wit four - but two of them in a single core 3GHz machine worked flawlessly doing only switching and IVR without codec conversion. Many will suggest that you split your lines on two machines to to prevent a total loss when a machine fails. This will add some work on setup but maybe save you some worries. Christian 2010/2/22 Arjan Kroon | Mobillion arjan.kr...@mobillion.nl: Hi, Does anybody have any experience with asterisk where are four PCIe cards are used in one server (TE420). So you can have max 4 * 4 * 30 channels = 480 channels used. Regards, Arjan Kroon Mobillion BV -- _ -- 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] 4 PCIe cards in one asterisk server
Hi, We are now using 2 PCI cards (TE410) in all our server without any problem. Because we want to reduce the power consumention of the complete server-park, we though to put 4 PCIe cards in 1 server. We have a redundancy of our servers, so machine fails is not a great issue. Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Christian Victor Verzonden: 22-02-2010 15:22 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] 4 PCIe cards in one asterisk server Not wit four - but two of them in a single core 3GHz machine worked flawlessly doing only switching and IVR without codec conversion. Many will suggest that you split your lines on two machines to to prevent a total loss when a machine fails. This will add some work on setup but maybe save you some worries. Christian 2010/2/22 Arjan Kroon | Mobillion arjan.kr...@mobillion.nl: Hi, Does anybody have any experience with asterisk where are four PCIe cards are used in one server (TE420). So you can have max 4 * 4 * 30 channels = 480 channels used. Regards, Arjan Kroon Mobillion BV -- _ -- 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] AMI Originate differences between 1.4 and 1.6.1
I do not remember and issues we have between 1.4 and 1.6. When going to your pastebin I get this: Sorry, an error has occurred. Reason: That is an invalid ID, or the post has expired. Can you post what your ami packets contain? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Feb 22, 2010, at 3:50 AM, Ritesh A wrote: Folks, I am strugging with Asterisk 1.4 Vs 1.6 differences over AMI Originate? Here is the pastebin... http://pastebin.ca/1805594 Not sure why the local channel won't send to context while the remote channel does. Worked fine in 1.4 but 1.6.1 has issues. Any help? Ritesh -- _ -- 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] 4 PCIe cards in one asterisk server
On Mon, Feb 22, 2010 at 8:20 AM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: Hi, Does anybody have any experience with asterisk where are four PCIe cards are used in one server (TE420). So you can have max 4 * 4 * 30 channels = 480 channels used. I would recommend calling Digium and asking them. They may have particular models that are known to work in that configuration. -- _ -- 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] HFC-S card
On 2/22/2010 1:02 PM, Tzafrir Cohen wrote: On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote: On 2/22/2010 10:26 AM, Per Jessen wrote: Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if that helps. /Per Jessen, Zürich I have use this howto http://www.voip-info.org/wiki/view/Asterisk+zaphfc; , but i can´t put the card working in nt ptp mode. Can you explain me how i have to do that? Do you have any howto to make the card work in nt ptp mode? thanks for answer Short answert: signalling = bri_net Longer answer: That page is outdated (hmm, and I didn't get to update it :-( ) Nowadays (as of Asterisk 1.6.0) BRI support is included in Asterisk. The zaphfc driver, though, is still not included in DAHDI. It's maintained, though. The version included in the Debian packages is taken from http://git.tzafrir.org.il/?p=dahdi-extra.git;a=summary . Either way (bristuff or Asterisk= 1.6.0) to use BRI PTP NT in chan_dahdi you should set: signalling = bri_net for the span's channels in /etc/asterisk/chan_dahdi.conf . I´m using centos 4.8 server, and i don't now how integrate zaphfc with dadhi -- _ -- 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] AMI Originate differences between 1.4 and 1.6.1
Here is the entire thing including problem statement, CLI, and AMI responses. http://pastebin.ca/1805792 Ritesh On Mon, Feb 22, 2010 at 8:31 PM, Jim Dickenson dicken...@cfmc.com wrote: I do not remember and issues we have between 1.4 and 1.6. When going to your pastebin I get this: Sorry, an error has occurred. Reason: *That is an invalid ID, or the post has expired.* * * Can you post what your ami packets contain? -- Jim Dickenson mailto:dicken...@cfmc.com dicken...@cfmc.com CfMC http://www.cfmc.com/ On Feb 22, 2010, at 3:50 AM, Ritesh A wrote: Folks, I am strugging with Asterisk 1.4 Vs 1.6 differences over AMI Originate? Here is the pastebin... http://pastebin.ca/1805594 Not sure why the local channel won't send to context while the remote channel does. Worked fine in 1.4 but 1.6.1 has issues. Any help? Ritesh -- _ -- 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] Caller ID question
Hiya - quick question.. When an external call is answered by an extension and the person answering the call wants to forward it to a different extension, is there any way to change the caller ID when the call is transferred? If someone is transferring a call to me, I see the caller ID of the other person in the office. When the call is transferred, could the caller ID be set back to the caller ID of the original incoming call? Staff members here often want to see the number of the last person they spoke to but when they check the call history on the (snom) phone, all they can see is the extension of the person that forwarded the call to them.. I doubt it's possible but thought I'd check Thanks, Will -- _ -- 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] Virtual machine timing (KVM)
On Sun, Feb 21, 2010 at 10:04 PM, Sean Brady sbr...@gtfservices.com wrote: I do get choppy audio when playing recordings occasionally. I haven’t had time to figure that one out, but I haven’t put it into production yet. You just said you're getting unexplained choppiness. You also just said you're not in production. I have been told repeatedly that Asterisk shouldn’t be virtualized, and that timing was an issue, however I have never been given a reason that I consider acceptable to preclude me from doing so. How about the fact you're getting unexplained choppiness before you're even in production? surrounding Asterisk virtualized. Perhaps I am just stubborn, but I am determined to run Asterisk virtualized in production with conferencing (be it meetme or confbridge) until it’s been proven without doubt that it just doesn’t work. What exactly would constitute 'proof without a doubt' that would satisfy you? If your virtualized webserver has to fight it out with other virts, and your webserver takes an extra second to process a web page, not such a big deal. If that's your audio conference that just had to spin for a second, you just lost words out of a sentence. If it happens during authentication, you dropped digits and the auth fails. If it happens during call setup, the call might not go through. If it happens during hangup, the hangup might get missed. UDP does NOT retransmit. Get it? Now do you understand why it's a bad idea? Timers are built on the premise that they have access to either a real timing device, or unobstructed access to a processor which clicks through a proc cycle at a pre-determined rate. Once you break those rules, don't be surprised when the timers stop working, and 'bad things' happen. -- _ -- 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 extensions
On 02/20/2010 01:53 AM, jonas kellens wrote: I have read on this list that people do not get a reply if they ask stupid questions. Is this then a stupid question that I ask ? If nobody has ever combined extensions.conf and realtime in a way that I want to do, I wanna hear it too. Even if this means no solution for me. Then I know it's not doable. Jonas. On Thu, 2010-02-18 at 20:15 +0100, jonas kellens wrote: How about something like : [mycontext] exten = 100,1,NoOp(calling 100) exten = 100,n,NoOp(going realtime) switch = Realtime/mycont...@realtime_extensions mailto:mycont...@realtime_extensions ; from here on we use realtime And then my MySQL-DB contains : `extensions_table` VALUES (1, 'mycontext', '100', n, 'Wait', '2'); `extensions_table` VALUES (2, 'mycontext', '100', n, 'NoOp', 'into RealTime'); 'extensions_table` VALUES (3, 'mycontext', '100', n, 'Playback', 'my-sound-file'); extconfig.conf has : realtime_extensions = mysql,asterisk,extensions_table Is all the above correct and possible ?? On Thu, 2010-02-18 at 13:55 -0500, Jared Smith wrote: On Thu, 2010-02-18 at 19:46 +0100, jonas kellens wrote: Does a context need completely be written or in extensions.conf or in the mysql-table 'extensions_table' ? Or can I combine the two with the 'switch'-statement ?? You can certainly combine the two with a switch statement. Asterisk will then only look in the switch if it doesn't find a match in extensions.conf. -- Jared Smith Digium, Inc. You can't use the n priority construct in realtime. the database schema won't tolerate it and has already been mentioned even if ti would it lacks the structure of the flat acsii file -- _ -- 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] Caller ID question
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten = s,1,set(callerid(num)=${passcallID})
Exten = s,n,dial(SIP/123)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Monday, February 22, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Caller ID question
Hiya - quick question..
When an external call is answered by an extension and the person answering
the call wants to forward it to a different extension, is there any way to
change the caller ID when the call is transferred?
If someone is transferring a call to me, I see the caller ID of the other
person in the office. When the call is transferred, could the caller ID be
set back to the caller ID of the original incoming call? Staff members here
often want to see the number of the last person they spoke to but when they
check the call history on the (snom) phone, all they can see is the
extension of the person that forwarded the call to them..
I doubt it's possible but thought I'd check
Thanks,
Will
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Re: [asterisk-users] Virtual machine timing (KVM)
David Backeberg wrote: Timers are built on the premise that they have access to either a real timing device, or unobstructed access to a processor which clicks through a proc cycle at a pre-determined rate. Once you break those rules, don't be surprised when the timers stop working, and 'bad things' happen. Forgive the possibly stupid question, but do these problems you describe apply equally to the dom0 as to any domU's in a xen system? I used to think not, but now I'm starting to realize that I'm probably mistaken... -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] Polycom VVX1500 video working yet?
On 19 February 2010 15:28, Steve Davies davies...@gmail.com wrote: [snip] I just upgraded to the new bootblock and 3.2.2 firmware, and these phones will now talk video to other devices. Nothing in the changelogs indicates why, but there is a definite jump up from the previous release of this phone. So, I duly stand corrected. OTOH, once upgraded, DHCP seems to be broken on these phones where static IP configuration is working perfectly :( I now have to eat my words a second time. Turns out that ISC's 3.1.1 dhcp server has a bug that is triggered by the way the VVX1500 requests its IP address. Upgrade to 3.1.3 and the VVX no longer has an issue with the DHCP server. In summary, the Polycom VVX 1500 is a very nice (if expensive) piece of kit, which seems to be playing nicely with Asterisk so far. 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] Caller ID question
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten = s,1,set(callerid(num)=${passcallID})
Exten = s,n,dial(SIP/123)
I thought about doing something like that but it would confuse the poor staff :)
They'd have a call from what appeared to be an external number but it would
turn out to be an internal extension that was calling them (we generally don't
blind transfer).
I need to change the CID on an already-established SIP channel and have no idea
if it's doable..
W
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Re: [asterisk-users] Virtual machine timing (KVM)
On Mon, Feb 22, 2010 at 10:51 AM, Jonathan Addleman j...@redowl.ca wrote: David Backeberg wrote: Timers are built on the premise that they have access to either a real timing device, or unobstructed access to a processor which clicks through a proc cycle at a pre-determined rate. Once you break those rules, don't be surprised when the timers stop working, and 'bad things' happen. Forgive the possibly stupid question, but do these problems you describe apply equally to the dom0 as to any domU's in a xen system? I used to think not, but now I'm starting to realize that I'm probably mistaken... http://wiki.xensource.com/xenwiki/Scheduling It sounds like there are multiple ways to do scheduling in a Xen situation. The best way to avoid overloading the system is to deliberately underutilize the system, but then what's the point of virtualization? The supposed benefits of virtualization are power savings, and better utilization of existing resources. If you're using it for other reasons like a development environment, you'll probably be fine. To be clear, you may get away with virtualization and never run into any problems. But you have to know who to blame when you DO run into problems. Having problems of the sort uniquely caused by starving virtual kernels for resources is not going to be the fault of asterisk, but rather a failure to anticipate the downside of trying to use virtualization with asterisk. -- _ -- 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] Caller ID question
The ID at dial/transfer time is what you are stuck with.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Monday, February 22, 2010 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID question
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten = s,1,set(callerid(num)=${passcallID})
Exten = s,n,dial(SIP/123)
I thought about doing something like that but it would confuse the poor
staff :)
They'd have a call from what appeared to be an external number but it would
turn out to be an internal extension that was calling them (we generally
don't blind transfer).
I need to change the CID on an already-established SIP channel and have no
idea if it's doable..
W
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Re: [asterisk-users] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM
Jason, Thanks for that, but I am still getting an error. I run rpmbuild using this command rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec --target=i686 --define kversion `uname -r` but it fails with this error message. make[1]: Leaving directory `/usr/src/kernels/2.6.18-128.el5-i686' + popd ~/localrpms/BUILD/dahdi-linux-kmod-2.2.1 + for kvariant in '' xen PAE + pushd _kmod_build_xen ~/localrpms/BUILD/dahdi-linux-kmod-2.2.1/_kmod_build_xen ~/localrpms/BUILD/dahdi-linux-kmod-2.2.1 + make KVERS=2.6.18-128.el5xen modules You do not appear to have the sources for the 2.6.18-128.el5xen kernel installed. make: *** [modules] Error 1 error: Bad exit status from /var/tmp/rpm-tmp.78040 (%build) I have installed the devel packages rpm -qa | grep kernel-.*devel kernel-xen-devel-2.6.18-128.el5 kernel-PAE-devel-2.6.18-128.el5 kernel-devel-2.6.18-128.el5 Is there something else I can do? Steve Hindmarch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: 16 February 2010 18:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM stephen.hindma...@bt.com wrote: rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec snip error: Failed build dependencies: kernel-devel = 2.6.18-164.11.1.el5 is needed by dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.i386 Add a --target=i686 to your rpmbuild line. -- _ -- 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] [OT] Asterisk 1.6 and DECT Phones
Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID question
On 22 February 2010 15:59, Will Payne w...@teambadger.co.uk wrote:
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten = s,1,set(callerid(num)=${passcallID})
Exten = s,n,dial(SIP/123)
I thought about doing something like that but it would confuse the poor staff
:)
They'd have a call from what appeared to be an external number but it would
turn out to be an internal extension that was calling them (we generally
don't blind transfer).
I need to change the CID on an already-established SIP channel and have no
idea if it's doable..
W
--
I believe what you want is called COLP Connected Line Presentation.
I was also if the opinion that it had been merged into all of the
newer versions of the Asterisk code.
If you are using Asterisk 1.4, you may find a usable patch here:
https://issues.asterisk.org/view.php?id=8824
Regards,
Steve
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Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. We've been using the Siemens Gigaset range for a few years now (specifically C475IP and S685IP). Not had any major problems with them. Regards, Chris -- For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons -- _ -- 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] Asterisk 1.6 and DECT Phones
On Mon, Feb 22, 2010 at 5:18 PM, --[ UxBoD ]-- ux...@splatnix.net wrote: Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil RTX3080 is a very good sip-dect product. The phones are really robust. And they have inferior battery time. The base station has also support for individual sip accounts per phone. I don't know where to get it globally. We import 8 at a time from a Danish distributor. Regards, Håkon -- _ -- 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] Virtual machine timing (KVM)
On Mon, Feb 22, 2010 at 11:06 AM, David Backeberg dbackeb...@gmail.comwrote: On Mon, Feb 22, 2010 at 10:51 AM, Jonathan Addleman j...@redowl.ca wrote: David Backeberg wrote: Timers are built on the premise that they have access to either a real timing device, or unobstructed access to a processor which clicks through a proc cycle at a pre-determined rate. Once you break those rules, don't be surprised when the timers stop working, and 'bad things' happen. Forgive the possibly stupid question, but do these problems you describe apply equally to the dom0 as to any domU's in a xen system? I used to think not, but now I'm starting to realize that I'm probably mistaken... http://wiki.xensource.com/xenwiki/Scheduling It sounds like there are multiple ways to do scheduling in a Xen situation. The best way to avoid overloading the system is to deliberately underutilize the system, but then what's the point of virtualization? The supposed benefits of virtualization are power savings, and better utilization of existing resources. If you're using it for other reasons like a development environment, you'll probably be fine. To be clear, you may get away with virtualization and never run into any problems. But you have to know who to blame when you DO run into problems. Having problems of the sort uniquely caused by starving virtual kernels for resources is not going to be the fault of asterisk, but rather a failure to anticipate the downside of trying to use virtualization with asterisk. -- There may be a way to use the Sangoma Voicetime USB timing device and map the Device to the VM. Its not possible in Citrix Xen but is possible in VMWare. -- _ -- 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] Asterisk 1.6 and DECT Phones
On 22 February 2010 16:18, --[ UxBoD ]-- ux...@splatnix.net wrote: Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. -- Thanks, Phil We use the snom M3s without any great difficulty, but the 1st thing to do when they come out of the box is to upgrade the firmware. Some of the earlier firmware revisions make these devices more useful as a doorstop! Additionally we use the Siemens Gigaset range. The only problems with them are 1) No auto configuration worth considering, and 2) Keeping up with the ever changing range. Again, be 100% sure to upgrade the firmware on these handsets when they come out of the box. 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] 4 PCIe cards in one asterisk server
I have always heard that the less cards in a single system the better. Why not try two Sangoma A108DE cards (8 ports each). Also make sure you have hardware echo cancellation on the cards for this number of ports. On Mon, 2010-02-22 at 14:20 +0100, Arjan Kroon | Mobillion wrote: Hi, Does anybody have any experience with asterisk where are four PCIe cards are used in one server (TE420). So you can have max 4 * 4 * 30 channels = 480 channels used. Regards, Arjan Kroon Mobillion BV -- _ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio to remote AGI server
On Monday 22 February 2010 03:49:48 Olle E. Johansson wrote:
22 feb 2010 kl. 07.23 skrev Tilghman Lesher:
open audio {tcp|udp} hostname portno
close audio
If you design something now, I would strongly suggest that we stop using
audio as an attribute. Each call will have multiple media streams - and
already have. You need to be able to select which one, and possibly open
multiple streams - audio, video, fax, text. In the future, we'll hopefully
have the ability to run multiple of each category, so I would not design
this feature for a single audio stream to be open for future use.
I doubt we'll support multiple streams of a similar type anytime in the near
future, and this is needed now. However, I'll additionally note that the
syntax I chose very easily can be adapted to open video, open text, etc.
--
Tilghman
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Re: [asterisk-users] string length in dialplan
Jerry Geis wrote:
I am trying to find out how I can tell the length of a string actually
CALLERID(num) in the dialplan.
How is that done?
If need to test the length of the CALLERID(num) if its less the 10 digits I
need to set it to a known value or insert 0's at the beginning until it
is 10 digits in length.
My PRI provider needs it set to 10 digits always.
...stuff before...
exten = _NXXNXX,n,GoSub(set_cid,1())
...stuff after...
exten = set_cid,1,NoOp()
exten = set_cid,n,Set(CURRENT_CID_LENGTH=${LEN(${CALLERID(num)})})
exten = set_cid,n,GotoIf($[${CURRENT_CID_LENGTH} = 10]?skip_modify_cid)
exten = set_cid,n,While($[${LEN(${CALLERID(num)})} 10])
exten = set_cid,n,Set(CALLERID(num)=0${CALLERID(num)})
exten = set_cid,n,EndWhile()
exten = set_cid,n(skip_modify_cid),Return()
There is likely a more efficient way of doing that, but I haven't gone through
and looked at the functions to see if there might be a way of avoiding the loop
:)
Also, totally untested, I just wrote it in this email ;)
Leif Madsen.
http://leifmadsen.wordpress.com
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Re: [asterisk-users] HFC-S card
Tzafrir Cohen wrote: On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote: On 2/22/2010 10:26 AM, Per Jessen wrote: Pedro Santos wrote: Does any one put a HFC-S card working in nt ptp mode? I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if that helps. /Per Jessen, Zürich I have use this howto http://www.voip-info.org/wiki/view/Asterisk+zaphfc; , but i can´t put the card working in nt ptp mode. Can you explain me how i have to do that? Do you have any howto to make the card work in nt ptp mode? thanks for answer Short answert: signalling = bri_net Longer answer: That page is outdated (hmm, and I didn't get to update it :-( ) Nowadays (as of Asterisk 1.6.0) BRI support is included in Asterisk. The zaphfc driver, though, is still not included in DAHDI. It's maintained, though. The version included in the Debian packages is taken from http://git.tzafrir.org.il/?p=dahdi-extra.git;a=summary . Either way (bristuff or Asterisk = 1.6.0) to use BRI PTP NT in chan_dahdi you should set: signalling = bri_net for the span's channels in /etc/asterisk/chan_dahdi.conf . None of the above looks very familiar - I'm using Asterisk 1.4.x + misdn, one HFC-4S for the external lines and one plain (Conrad) HFS-PCI in NT mode for an ISDN DECT base-station. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. -- _ -- 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] Multiple instances of Asterisk on the same host...
Interesting thread recently about virtual servers... I'm thinking of doing something similar - right now looking at Containers (lxc) rather than proper virtualisation though, however it got me thinking of a poor mans virtualisation solution... This would assume you have a real server to start with and full root access... I was thinking of simply running multiple asterisks on the same box, each with their own /etc/asterisk config directory (in e.g. /home/v1/etc/asterisk, /home/v2/etc/asterisk and so on - obviously give them unique /home/v1/spool/asterisk/ , etc. directories too, but for the most part things like /var/lib/asterisk/sounds and modules can be shared. (exception being astdb!) It just means a custom /etc/asterisk/asterisk.conf file for each instance and asterisk being started with the correct config file - /home/v1/etc/asterisk.conf, etc. So giving each asterisk it's own IP address (eth0:1, eth0:2, etc.) and changing the bindaddr parameter in each one to suit multiple IP addresses bound to the 'host' would seem to be the way to do it - each asterisk can still use ztdummy/dhadidummy for timing if required (or does it stop multiple asterisks opening it?) Anyone done this or contemplated doing it? 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] [OT] Asterisk 1.6 and DECT Phones
Running Asterisk trunk with Siemens Gigaset S685IP, no normal problems, just some with connected-line, probaly me, who is not smart enough. :( Sound is great, use them both at our WAN and NAT'et at my home, DTMF working as a clock... what more can I say? On Mon, 22 Feb 2010 16:43:04 -, Chris Bagnall wrote: looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. We've been using the Siemens Gigaset range for a few years now (specifically C475IP and S685IP). Not had any major problems with them. Regards, Chris -- For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons -- _ -- 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] string length in dialplan
On Mon, Feb 22, 2010 at 12:57:30PM -0500, Leif Madsen wrote:
Jerry Geis wrote:
I am trying to find out how I can tell the length of a string actually
CALLERID(num) in the dialplan.
How is that done?
If need to test the length of the CALLERID(num) if its less the 10 digits I
need to set it to a known value or insert 0's at the beginning until it
is 10 digits in length.
My PRI provider needs it set to 10 digits always.
...stuff before...
exten = _NXXNXX,n,GoSub(set_cid,1())
...stuff after...
exten = set_cid,1,NoOp()
exten = set_cid,n,Set(CURRENT_CID_LENGTH=${LEN(${CALLERID(num)})})
exten = set_cid,n,GotoIf($[${CURRENT_CID_LENGTH} = 10]?skip_modify_cid)
exten = set_cid,n,While($[${LEN(${CALLERID(num)})} 10])
exten = set_cid,n,Set(CALLERID(num)=0${CALLERID(num)})
exten = set_cid,n,EndWhile()
exten = set_cid,n(skip_modify_cid),Return()
There is likely a more efficient way of doing that, but I haven't gone
through
and looked at the functions to see if there might be a way of avoiding the
loop :)
His provider wants 10 digits always, so
exten = set_cid,n,Set(FOO=00${CALLERID(num)})
exten = set_cid,n,Set(CALLERID(num)=${FOO:-10})
would work, but in that case he's likely going to present annoying CIDs
like 000666 to his callees.
--
Barry
--
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Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
On Mon, 22 Feb 2010, --[ UxBoD ]-- wrote: Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. Siemens Gigaset over M3's anyday. Nicer displays, bigger handsets and buttons. The downside is that they are slow - both on the handsets and their web interface - make sure you're using a browser with fast javascript - e.g. Chrome. 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] [OT] Asterisk 1.6 and DECT Phones
- Gordon Henderson gordon+aster...@drogon.net wrote: On Mon, 22 Feb 2010, --[ UxBoD ]-- wrote: Hi, looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. Siemens Gigaset over M3's anyday. Nicer displays, bigger handsets and buttons. The downside is that they are slow - both on the handsets and their web interface - make sure you're using a browser with fast javascript - e.g. Chrome. Gordon I did buy a Gigaset and subsequently sold it on due to its speed :( I would love to continue using Snom but they do not appear to have done any testing with 1.6 branch and my phones keep going lagged. Reading the forums about recent issues would make me cry as a manufacturer! -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple instances of Asterisk on the same host...
Gordon Henderson wrote: Interesting thread recently about virtual servers... I'm thinking of doing something similar - right now looking at Containers (lxc) rather than proper virtualisation though, however it got me thinking of a poor mans virtualisation solution... This would assume you have a real server to start with and full root access... I was thinking of simply running multiple asterisks on the same box, each with their own /etc/asterisk config directory (in e.g. /home/v1/etc/asterisk, /home/v2/etc/asterisk and so on - obviously give them unique /home/v1/spool/asterisk/ , etc. directories too, but for the most part things like /var/lib/asterisk/sounds and modules can be shared. (exception being astdb!) It just means a custom /etc/asterisk/asterisk.conf file for each instance and asterisk being started with the correct config file - /home/v1/etc/asterisk.conf, etc. So giving each asterisk it's own IP address (eth0:1, eth0:2, etc.) and changing the bindaddr parameter in each one to suit multiple IP addresses bound to the 'host' would seem to be the way to do it - each asterisk can still use ztdummy/dhadidummy for timing if required (or does it stop multiple asterisks opening it?) Anyone done this or contemplated doing it? I have heard of a company, name completely escapes me right now, that appears to use Linux-Vserver. I am trying to find the time to move my business system to a Linux-Vserver from a Micro-Linux Asterisk Server and the only issue I'm aware of is DAHDI/ZAPTEL might have to be run in the host instead of the guests. Then some permissions set so the guests can access it DAHDI. \\||/ Rod -- 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
[asterisk-users] init.d error when installing trunk
Hi, The last few times I have installed trunk versions of asterisk on Ubuntu I have seen this error after doing a make config for asterisk. install: cannot stat `contrib/init.d/etc_default_asterisk': No such file or directory The init.d links then fail to work properly (e.g. /etc/init.d/asterisk restart) after installation. Most recently I installed asterisk (SVN-trunk-r248269) on Ubuntu Server 9.10. Have I missed something in the install process somewhere? Thanks, Nic. -- _ -- 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] Open source or low-budget recommendation for call-center software
Hello, We used to recommend a commercial software but client is a small callcenter who cannot afford something big. Would you recommend something open-source which could work for a 40-seater? Thank you, Tudor www.sunabasarabia.com Moldova 11c/min Romania 2c/min $1 de test de la bun inceput -- _ -- 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] Problems with SIP realtime
I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from 'sip:test...@192.168.1.150;transport=UDP' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/AsteriskHosted/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/AsteriskHosted/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/AsteriskHosted/sip_buddies I have the following in extconfig.conf : sipusers = mysql,Asterisk,sip_buddies sippeers = mysql,Asterisk,sip_buddies I have the following in res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 dbsock = /tmp/mysql.sock Something I'm missing ?? Need extra configuration ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem w/ MoH
Hi all, I'm trying to get moh working on * version 1.4.4. I've setup a test extension that answers the call and runs the musiconhold command with the appropriate class name. All I get on the phone is silence. The console tells me that moh started and immediately stopped, but it complains that there is No class: moh0 *CLI [Feb 22 12:17:36] WARNING[31142]: res_musiconhold.c:947 local_ast_moh_start: No class: (moh0) Here is the appropriate output of moh show classes Class: moh0 Mode: custom Directory: /etc/asterisk/diehl/music/moh0/ Application: /usr/bin/madplay -Q -z ---mono -R 8000 -o raw:- -r -a -12 Format: slin I have confirmed that madplay is in /usr/bin/. I've also used the playback command to ensure that the .wav files in the moh0 directory can be played by Asterisk. What am I missing? TIA, 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] Problems with SIP realtime
Little fault in my mailing : The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/Asterisk/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/Asterisk/sip_buddies My database-name is just 'Asterisk', my bad. So... what am I missing for this realtime SIP to work ?? Jonas On Mon, 2010-02-22 at 20:13 +0100, jonas kellens wrote: I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from 'sip:test...@192.168.1.150;transport=UDP' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/Asterisk/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/Asterisk/sip_buddies I have the following in extconfig.conf : sipusers = mysql,Asterisk,sip_buddies sippeers = mysql,Asterisk,sip_buddies I have the following in res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 dbsock = /tmp/mysql.sock Something I'm missing ?? Need extra configuration ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem w/ MoH
On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote: Hi all, I'm trying to get moh working on * version 1.4.4. I've setup a test I don't know the answer, but are you really using 1.4.4? If so, consider taking some time to review the security and feature improvements over the last several years and seriously consider updating. -- _ -- 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] Problem w/ MoH
David Backeberg wrote: On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote: Hi all, I'm trying to get moh working on * version 1.4.4. I've setup a test I don't know the answer, but are you really using 1.4.4? If so, consider taking some time to review the security and feature improvements over the last several years and seriously consider updating. I am planning to upgrade, but first, I need everything working. Then I need to upgrade my backup system and test/fix. then I'll move my customers. Kinda drawn out, I know. 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] Problems with SIP realtime
The problem was that I had a different value for 'name' and 'username'. How can I have the 'name' different from the 'username' ??? Why do these 2 need to be the same ?? Jonas. On Mon, 2010-02-22 at 20:36 +0100, jonas kellens wrote: Little fault in my mailing : The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/Asterisk/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/Asterisk/sip_buddies My database-name is just 'Asterisk', my bad. So... what am I missing for this realtime SIP to work ?? Jonas On Mon, 2010-02-22 at 20:13 +0100, jonas kellens wrote: I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from 'sip:test...@192.168.1.150;transport=UDP' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/Asterisk/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/Asterisk/sip_buddies I have the following in extconfig.conf : sipusers = mysql,Asterisk,sip_buddies sippeers = mysql,Asterisk,sip_buddies I have the following in res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 dbsock = /tmp/mysql.sock Something I'm missing ?? Need extra configuration ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with SIP realtime
Hello Jonas: Change this parameter, if you are using Mysql. [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 *dbsock = /var/lib/mysql/mysql.sock* cheersss... 2010/2/22 jonas kellens jonas.kell...@telenet.be The problem was that I had a different value for 'name' and 'username'. How can I have the 'name' different from the 'username' ??? Why do these 2 need to be the same ?? Jonas. On Mon, 2010-02-22 at 20:36 +0100, jonas kellens wrote: Little fault in my mailing : The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/Asterisk/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/Asterisk/sip_buddies My database-name is just 'Asterisk', my bad. So... what am I missing for this realtime SIP to work ?? Jonas On Mon, 2010-02-22 at 20:13 +0100, jonas kellens wrote: I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from 'sip:test...@192.168.1.150sip%3atest...@192.168.1.150;transport=UDP' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/Asterisk/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/Asterisk/sip_buddies I have the following in extconfig.conf : sipusers = mysql,Asterisk,sip_buddies sippeers = mysql,Asterisk,sip_buddies I have the following in res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 dbsock = /tmp/mysql.sock Something I'm missing ?? Need extra configuration ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Avaya with Asterisk
I have a connection of Asterisk with Avaya by H.323 and so far everything worked well because only sent to Avaya. Now, the matter is that from Avaya will send me an IVR calls to capture credit card information, the link is active on Avaya 23 channels which is not how to configure Asterisk for those 23 simultaneous channels of Avaya's collect asterisk. Do not know if I can be with a group or queue, the idea is that all calls go to one place and who answer all calls is the IVR. Any suggestions or ideas? Edwin Quijada *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] Problems with SIP realtime
Dear Juan, thank you for your answer. The reason why registration failed was a mismatch between the 'username'-field and the 'name'-field. If I put both values to the same, it works... But why do these 2 need to be the same ? I would rather have a different 'name' and 'username'-parameter. Adding 'authname' to my realtime MySQL-DB does not change anything. (found this field 'authname' through google) Jonas. On Mon, 2010-02-22 at 15:25 -0500, Juan Miguel wrote: Hello Jonas: Change this parameter, if you are using Mysql. [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 dbsock = /var/lib/mysql/mysql.sock cheersss... -- _ -- 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] SIP Disconnects from Network - Asterisk Does not hangup
Good day all! I have an issue which has plagued me for quite sometime now...and as I close in on its cause, I have reached a point where additional info would be greatly helpful! When a SIP device dials another SIP device...Asterisk connects the calls and displays the channel information. If one of those SIP devices hangs up, Asterisk receives the hangup notice and disconnects the call/channel. However - what does Asterisk do when the network cable is unplugged from one of the SIP devices...?! From what I'm seeing here, it does nothing! Asterisk still shows the call as being live and only reports that the SIP device has become unreachable (in full log). Is this something that is fixed in an update? (Currently running 1.2) It seems when Asterisk detects the SIP device has become unresponsive, it would auto-disconnect any calls bridged to that device...though its not. Thus creating what I like to call 'Phantom Calls'. I can arrive in the morning and view calls that have gone longer than 24 hours - even though those callers hung up many hours prior. -- _ -- 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] Open source or low-budget recommendation for call-center software
I think Vicidial, works great. Regards. 2010/2/22 Apa Minerala apaminer...@yahoo.com Hello, We used to recommend a commercial software but client is a small callcenter who cannot afford something big. Would you recommend something open-source which could work for a 40-seater? Thank you, Tudor www.sunabasarabia.com Moldova 11c/min Romania 2c/min $1 de test de la bun inceput -- _ -- 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 -- Juan. -- _ -- 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] Open source or low-budget recommendation for call-center software
GnuDialer *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: juanch...@gmail.com Date: Mon, 22 Feb 2010 16:37:22 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Open source or low-budget recommendation for call-center software I think Vicidial, works great. Regards. 2010/2/22 Apa Minerala apaminer...@yahoo.com Hello, We used to recommend a commercial software but client is a small callcenter who cannot afford something big. Would you recommend something open-source which could work for a 40-seater? Thank you, Tudor www.sunabasarabia.com Moldova 11c/min Romania 2c/min $1 de test de la bun inceput -- _ -- 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 -- Juan. _ -- _ -- 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] Problems with SIP realtime
On 02/22/2010 11:13 AM, jonas kellens wrote: I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from 'sip:test...@192.168.1.150;transport=UDP' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22 19:58:23] Found [Feb 22 19:58:23] == Binding voicemail to mysql/AsteriskHosted/voicemail_users [Feb 22 19:58:23] == Binding sipusers to mysql/AsteriskHosted/sip_buddies [Feb 22 19:58:23] == Binding sippeers to mysql/AsteriskHosted/sip_buddies I have the following in extconfig.conf : sipusers = mysql,Asterisk,sip_buddies sippeers = mysql,Asterisk,sip_buddies I have the following in res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = Asterisk dbuser = asteriskuser dbpass = asteriskpasswd dbport = 3306 dbsock = /tmp/mysql.sock Something I'm missing ?? Need extra configuration ? Jonas. The error message would seem to indicate that the endpoint isn't being matched in sip_buddies. you need the host column to be set to dynamic for the peer. All else seems to be in order -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
On Mon, 2010-02-22 at 16:13 -0500, JT wrote: Is this something that is fixed in an update? (Currently running 1.2) Yes... modern versions of Asterisk support SIP session timers. (If I remember correctly, Asterisk 1.2 could tear down a call based on lack of RTP data, but I never found it worked well enough in my tests to warrant its use.) -- Jared Smith Digium, 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
[asterisk-users] Load balance outgoing calls
Hello everybody.
I have a provider that has 3 asterisk boxes which I must balance my
calls against. At the moment, I route different destinations to
different boxes but this causes lots of problems.
Without resorting to OpenSER or other proxies (as my provider also
uses IAX), is there a way I can load balance outgoing channels in
Asterisk?
For example an IAX peer like:
[iax_provider]
type=peer
username=myprovider
host=xxx.xxx.xxx.10
host=xxx.xxx.xxx.11
host=xxx.xxx.xxx.12
secret=verysecret
disalow=all
allow=g729
Is there any way I can balance calls between all of the hosts in the
providers description? In fact, if I set the dialplan like:
exten = _X.,n,Dial(IAX2/iax_provider/${EXTEN}
what IP addres will receive the call? host 10, 11 or 12?
I know DAHDI can balance outgoing calls between the E1's of the span
using DAHDI/r0/ instead of DAHDI/g0. Is there any way of doing this
for other channels?
Thanks!
Alex
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Re: [asterisk-users] Virtual machine timing (KVM)
I had a system running on Xen in test. I had terrible echo problems with a SPA3000. As a reference, I swapped to bare metal machine and although I still had echoing, the echoing was much closer to the original sound. The Xen server was idle apart from the AsteriskNOW installation. So, this lead me to believe that Xen was introducing some latency somewhere this could be due to bridging overheads or something... not necessarily due to processor starvation. I was really disappointed because it has taken away the whole viability of the project I was running with. I might get better luck with some better echo cancellation, but the latency introduced would still affect normal two way conversations. Running under Xen also had some interesting effects on DTMF tones, etc. -- _ -- 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] Virtual machine timing (KVM)
Forgive the possibly stupid question, but do these problems you describe apply equally to the dom0 as to any domU's in a xen system? I used to think not, but now I'm starting to realize that I'm probably mistaken... Dom0 is still a virtual machine, so I would say so. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf - sort order, does it matter
On 02/19/10 08:54, Olle E. Johansson wrote: 17 feb 2010 kl. 19.12 skrev Joseph: Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and extension.conf) insecure=invite is not taking effect. I get: ... username mismatch, have 4, digest has pstn- handle_request_invite: Failed to authenticate user KMIEC J You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. /O Yes, I have type=friend but I've loaded my other dial plan where I have type=friend as well and insecure=invite is working. So, it must be the sort order that is generating problems. -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple instances of Asterisk on the same host...
On Mon, 22 Feb 2010, Roderick A. Anderson wrote: Gordon Henderson wrote: Interesting thread recently about virtual servers... I'm thinking of doing something similar - right now looking at Containers (lxc) rather than proper virtualisation though, however it got me thinking of a poor mans virtualisation solution... This would assume you have a real server to start with and full root access... I was thinking of simply running multiple asterisks on the same box, each with their own /etc/asterisk config directory (in e.g. /home/v1/etc/asterisk, /home/v2/etc/asterisk and so on - obviously give them unique /home/v1/spool/asterisk/ , etc. directories too, but for the most part things like /var/lib/asterisk/sounds and modules can be shared. (exception being astdb!) It just means a custom /etc/asterisk/asterisk.conf file for each instance and asterisk being started with the correct config file - /home/v1/etc/asterisk.conf, etc. So giving each asterisk it's own IP address (eth0:1, eth0:2, etc.) and changing the bindaddr parameter in each one to suit multiple IP addresses bound to the 'host' would seem to be the way to do it - each asterisk can still use ztdummy/dhadidummy for timing if required (or does it stop multiple asterisks opening it?) Anyone done this or contemplated doing it? I have heard of a company, name completely escapes me right now, that appears to use Linux-Vserver. I am trying to find the time to move my business system to a Linux-Vserver from a Micro-Linux Asterisk Server and the only issue I'm aware of is DAHDI/ZAPTEL might have to be run in the host instead of the guests. Then some permissions set so the guests can access it DAHDI. My aim is to actually use LXC as it has kernel level support (as of 2.6.29) and will be supported by most distros soon if not already. Linux-Vserver appears to be depreciated by at least Debian, probably Ubuntu too, but I've no idea about the world of Red Hat/Fedora/Centos, etc.. I tried OpenVZ, but it seems to have even poorer support, and no updated for some time either. Using Containers (as opposed to virtualisation), I'd expect DAHDI/Zap to be run on the host as there's only one kernel. Each container looking like a super chrooted environment, but one which can still have it's own /sbin/init firing off local processes, network configurations, local filesystem, etc. 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] SIP Disconnects from Network - Asterisk Does not hangup
On 100222 1313, JT wrote: When a SIP device dials another SIP device...Asterisk connects the calls and displays the channel information. If one of those SIP devices hangs up, Asterisk receives the hangup notice and disconnects the call/channel. However - what does Asterisk do when the network cable is unplugged from one of the SIP devices...?! Jared already mentioned SIP session timers, which are supported starting with 1.6. Here's my experience. While I am running 1.6, the software stack that is used for agent softphone (PJSIP) does not support the session timers. If the softphone crashes in a call, the call would get stuck exactly as you describe. I am working around this problem by setting rtp timeouts in sip.conf: [general] rtptimeout=10 rtpholdtimeout=300 This means that if RTP flow stops while the agent is in the call, the call will be disconnected in 10 seconds. If the call was put on hold by the agent, it will be disconnected in 300 seconds. Your timeouts may vary. The caveat here is that it is perfectly normal NOT to transmit any RTP data in case of long silence. This is why the SIP timers were introduced in the first place: there is no correct way to detect when the client is going away, as no activity is a good session state. I am able to get away with the small timeout because I set the PJSIP client to always transmit RTP, by turning off voice activity detection feature (VAD). If you want to support that feature, set rtptimeout as high as for how long you allow absolute silence on the line without disconnecting it. I do not know if these settings are available in 1.2 though. -kkm smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balance outgoing calls
On Mon, 22 Feb 2010, Alejandro Recarey wrote:
I have a provider that has 3 asterisk boxes which I must balance my
calls against. At the moment, I route different destinations to
different boxes but this causes lots of problems.
[snip]
Is there any way I can balance calls between all of the hosts in the
providers description?
Name the provider hosts something like isp0, isp1, and isp2. Then in
extensions.conf, use something like:
dial(iax2/isp${MATH(${EPOCH}%3):0:1}/${EXTEN})
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] TE410P Spans offline/red after power down/restart
On Mon, Feb 22, 2010 at 6:13 PM, Benoit maver...@maverick.eu.org wrote: There is one similar request on this list from a few weeks back iirc Oh, I'd looked through the archives/googled etc. but could not find anything similar. I'll take another stab at the archives. We quickly removed the card and checked the jumpers, suspecting that it may somehow have 'changed' it's state. Checking the card showed that all ports were open, which is what we expected. Re-setting the card done nothing to resolve our problem. You did not disconnect cables by doing so ? How much time did you let the card disconnected from everything, at least 30s ? Oh, yes, when we removed the card the cables were of course disconnected. The issue *seems* to be when they are connected at power up. The card was removed from the server for probably close to 5 minutes. If i'm not misleading isdn lines are powered, not huge amount of power but still more than an ethernet cable. Maybe the power was sufficient to held the card in a misconfigured state I suspected that this may be the case, and perhaps the ports on our PBX were able to 're-configure' the line. We have another power down this weekend with the same setup, I will see if I can test this theory out then. Here are the versions we are running: Asterisk 1.4.18.1 Zaptel 1.4.9.2-48 Libpri 1.4.3-19 Or you are simply using very outdated drivers/software :) I'm sure that could very well be the casebut, unfortunately we are tied to our current versions/platform -- _ -- 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] Load balance outgoing calls
On Mon, 22 Feb 2010, Steve Edwards wrote:
dial(iax2/isp${MATH(${EPOCH}%3):0:1}/${EXTEN})
Improving on myself...
Using the decimal portion of UNIQUEID (the number of channels
created by this instance of Asterisk) would be better than EPOCH.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Virtual machine timing (KVM)
Ian Murray wrote: Forgive the possibly stupid question, but do these problems you describe apply equally to the dom0 as to any domU's in a xen system? I used to think not, but now I'm starting to realize that I'm probably mistaken... Dom0 is still a virtual machine, so I would say so. Ok, thanks! Another stupid question that I probably know (or should know) the answer to: if all the other virtual machines are shut down, should the dom0 return to normal, or does simply having a xen-enabled kernel cause trouble? I imagine it's the actual sharing that's at fault, but who knows..? Thanks! -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] Asterisk 1.6 and DECT Phones
Hi! looking for your valued input on suitable suggestions for high quality VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking to a new manufacturer. Define high quality. Anyone here used any of these below with Asterisk? * NEC AP300 and NEC DECT C124 or NEC DECT M155 * Aastra RFP L32 with Aastra 142 DECT or Aastra 610d/620d DECT I am really curious about those, especially the M155. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
Kirill 'Big K' Katsnelson wrote: The caveat here is that it is perfectly normal NOT to transmit any RTP data in case of long silence. This is why the SIP timers were introduced in the first place: there is no correct way to detect when the client is going away, as no activity is a good session state. That's only true when Asterisk tells the other endpoint that it is allowed to use voice activity detection and silence suppression, which at this point it does not do. In spite of that, there are many endpoints that do it anyway, which then causes strange problems on calls, including calls getting dropped if an RTP timeout is in use. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@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] SIP Disconnects from Network - Asterisk Does not hangup
On 100222 1818, Kevin P. Fleming wrote: Kirill 'Big K' Katsnelson wrote: The caveat here is that it is perfectly normal NOT to transmit any RTP data in case of long silence. This is why the SIP timers were introduced in the first place: there is no correct way to detect when the client is going away, as no activity is a good session state. That's only true when Asterisk tells the other endpoint that it is allowed to use voice activity detection and silence suppression, which at this point it does not do. In spite of that, there are many endpoints that do it anyway, Oh yes, I've seen these problems first person, mostly manifesting themselves as dropped syllables after a period of silence if not complete loss of a call, but I assumed it was not a negotiated option but rather left to unilateral decision of an endpoint. Thank you for the correction! -kkm smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
23 feb 2010 kl. 03.18 skrev Kevin P. Fleming: Kirill 'Big K' Katsnelson wrote: The caveat here is that it is perfectly normal NOT to transmit any RTP data in case of long silence. This is why the SIP timers were introduced in the first place: there is no correct way to detect when the client is going away, as no activity is a good session state. That's only true when Asterisk tells the other endpoint that it is allowed to use voice activity detection and silence suppression, which at this point it does not do. In spite of that, there are many endpoints that do it anyway, which then causes strange problems on calls, including calls getting dropped if an RTP timeout is in use. Well, the headers we use are note really standardized, at least I could not find them. In the RTP rfc's it's perfectly legal to just have gaps in the timestamps and stop sending. However, as both me and Kevin stated, Asterisk does not support it. On most phones, you can disable silence suppression in the configuration. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson: On 100222 1313, JT wrote: When a SIP device dials another SIP device...Asterisk connects the calls and displays the channel information. If one of those SIP devices hangs up, Asterisk receives the hangup notice and disconnects the call/channel. However - what does Asterisk do when the network cable is unplugged from one of the SIP devices...?! Jared already mentioned SIP session timers, which are supported starting with 1.6. Here's my experience. While I am running 1.6, the software stack that is used for agent softphone (PJSIP) does not support the session timers. If the softphone crashes in a call, the call would get stuck exactly as you describe. I am working around this problem by setting rtp timeouts in sip.conf: [general] rtptimeout=10 rtpholdtimeout=300 This means that if RTP flow stops while the agent is in the call, the call will be disconnected in 10 seconds. If the call was put on hold by the agent, it will be disconnected in 300 seconds. Your timeouts may vary. The caveat here is that it is perfectly normal NOT to transmit any RTP data in case of long silence. Not in Asterisk - we do not really support silence suppression. The recommendation is to turn it off on the phones. This is why the SIP timers were introduced in the first place: there is no correct way to detect when the client is going away, as no activity is a good session state. I am able to get away with the small timeout because I set the PJSIP client to always transmit RTP, by turning off voice activity detection feature (VAD). If you want to support that feature, set rtptimeout as high as for how long you allow absolute silence on the line without disconnecting it. Just to complete this discussion - we also have the absolute timeout that is a lifesaver in many cases. If you set this to a time that's larger than the normal calls, Asterisk will hang up the call. I very often set it to two hours, just to make sure that if anything strange happens, all calls will be cancelled out at some point. /O -- _ -- 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
