Re: [asterisk-users] Fax Server for Asterisk
On 05/30/2012 12:02 AM, Danny Dias wrote: Hi all, Does Hylafax and IAXmodem works with analog lines? or only with E1? I've been checking some commercial solutions (in case Asterisk is not on site, and the customer wants a Fax Server as standalone), i saw FaxBack and Linkcom e-fax But again, if Hylafax and iaxmodem works also with analog lines, that would be better to use. Could you please confirm? any place to check How-To on Hylafax and Iaxmodem? not sure what your requirements are, but i can +1 the reliability of hylafax, have been using it for years problem free although only +- 10 faxes a day, and its running on an old leftover pc with a cheap modem on a analog line. i use it in combination with avantfax as a web front end and email2fax, its a cheap and Free solution, easy to use, low maintenance etc never used iaxmodem though Many thanks!!! 2012/5/29 Carlos Alvarez car...@televolve.com mailto:car...@televolve.com On Tue, May 29, 2012 at 8:03 AM, Warren Selby wcse...@selbytech.com mailto:wcse...@selbytech.com wrote: On Tue, May 29, 2012 at 3:10 AM, Danny Dias ing.diasda...@gmail.com mailto:ing.diasda...@gmail.com wrote: Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks I've been real happy with using HylaFax+ and Iaxmodem implementations. We have a few Hylafax servers in our network. Both it and IAXmodem are a real bear to learn at first (well, so is Asterisk) but when you get them working, they are rock solid. I hadn't even thought about it, but it's been at least a year since I logged into any of our Hylafax servers and did anything to them. They just work. I would estimate I put in a solid 30 hours into learning and configuring the first server, and then some more time learning additional capabilities and best practices. But again, since doing that, it's been totally hands-off. I will add though that we also use Fax for Asterisk simply to receive and turn faxes into PDF for some customers, and that is perfectly stable also. -- Carlos Alvarez TelEvolve 602-889-3003 tel:602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- www.danntel.net http://www.danntel.net _sip:danny4...@thesipschool.com mailto:danny4...@thesipschool.com_ sip:dann...@opensips.org mailto:sip%3adann...@opensips.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Group call from DAHDI
Hello, I am trying to figure out how to make group call using DAHDI. I want to make multiple call at once and conference among them. I know about meetme, but that is for incoming conference call. Please suggest. -- Regards, Ashish Agarwal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] axfer with simple CDR
Dne 29.5.2012 18:23, Kevin P. Fleming napsal(a): On 05/29/2012 07:57 AM, Marek Cervenka wrote: is it possible with simple CDR fully describe axfer? (axfer is asterisk native, not phone function) No, it is not. CDRs (Asterisk or otherwise) are only capable of directly (simply) describing a call from party A to party B. They have no ability to describe call treatments, in-call features, or any other advanced features. Asterisk's CDRs *attempt* to represent such information, but as you've seen, they don't satisfy everyone, and it seems that many parties have conflicting ideas as to how things like transfers should be represented in CDRs. ok ok. i tried it :) i'll try it the right way - CEL (centos6,unixODBC,cel_odbc,mysql) any sql views,scripts,sql triggers someone? is it implemented in switchvox,asterisknow,trixbox,elastix? -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Administrator TOOTAI wrote: 10.0.70.12 is the IP of Asterisk server (kvm virtual machine) which is replaced by externaddr parameter from sip.conf. If you have other ideas, welcome ;-) Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. This is confusing because your first email said you had nat=no in your working 1.6.24 setup, but everything you're saying indicates a NAT problem to me. A diagram showing all network elements between your Asterisk server and the remote host would be helpful. To avoid further confusion, please include full and unaltered logs, SIP traces, and configurations in future posts. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. It sounds like a NAT issue to me too. Why don't you do a quick test and put the Asterisk box on a public IP if you can. If it works, you will have narrowed down the issue to a NAT problem. You could have a nat router with a broken SIP ALG. -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Introducing Limesco
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Earlier this year I spoke to Malcolm Davenport from Digium about a mobile phone operator we're starting in The Netherlands. He really liked the idea and proposed to send an email to this mailing list. Since this message is kind of spam and a little off-topic, I'll limit myself to this single message and I also request any questions or replies to be kept of the list. To the point: in The Netherlands we're starting Limesco, a mobile phone operator aimed at IT specialists. What's unique about Limesco, is that we want to bring more control to the end user, compared to what's possible with existing operators. Our most interesting offer from the start is usage of our gateway that translates signaling and voice streams between one of the existing mobile networks in +31 and the world of SIP and RTP. Essentially, this allows you to use any GSM or 3G compatible device, insert a Limesco SIM and use it as if it were an extension on an Asterisk server. All calls are routed over SIP without running special software on the mobile phone. At this moment, we're finishing up some organizational stuff and we want to start running with a closed pilot group soon. Normal usage charges apply during this period, but Limesco likes to reimburse part of the activation and monthly fee for pilot users that have nice ideas for experimentation and that also document their results on our wiki. Since Limesco only operates in The Netherlands for now, most of our website and communication is in Dutch. However, if you're interested in the project or the pilot, please become a member of Vereniging Limesco (with legal voting rights in the organization) at https://limesco.org/ or directly contact me by email. Thank you, Mark van Cuijk -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.17 (Darwin) Comment: GPGTools - http://gpgtools.org iEYEARECAAYFAk/GO6QACgkQnfxiFjIAu8dyAwCfWSfq8ru3VhZ1KUqfvr2PHVid PyoAoOFXhOehiWeRp3b+pdf3pNSJTrxq =ZY+6 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loss of RTP stream during DTMF collection
On 05/25/2012 06:30 PM, Dave George wrote: How can I enable the option to allow asterisk to maintain the RTP stream during DTMF collection? If it's the problem I hypothesized it was, you can set 'transmit_silence=yes' in your asterisk.conf file. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Le 30/05/2012 14:44, Matthew J. Roth a écrit : Considering that you made progress on your initial problem by setting nat=force_rport (resulting in connected calls with no audio) and now you're mentioning the use of externaddr, I'd recommend a very careful reading of the NAT SUPPORT section of sip.conf.sample in the configs directory of the Asterisk source tree. I did read all those documentation, belive me. Also keep in mind that I *ONLY* face this problem with this provider, people using voipbuster or sipdiscount should have the same problem. Concerning externaddr, this test server -dedicated to asterisk- being running in VM since ages, I never would suspect a NAT issue! Especially if previous 1.4 and 1.6 version are running smoothly ... In Asterisk 1.8, there is a new configuration option named media_address which may be of particular interest. media_address seems not an option, can be set only in general not per peer. This is confusing because your first email said you had nat=no in your working 1.6.24 setup, but everything you're saying indicates a NAT problem to me. Again, 1.6 version is perfectly working with this setup and conf files, and before 1.4 was too. And those both asterisk versions with *this* provider. . A diagram showing all network elements between your Asterisk server and the remote host would be helpful. Phone registration: phone (Snom320 and GS GXV3175) - firewall1 (linux router) - Internet - firewall2 (linux router) - VM - phone account Call: phone account - Out of VM - firewall2 (linux router) - Internet - Peer IP - ??? To avoid further confusion, please include full and unaltered logs, SIP traces, and configurations in future posts. During the time you and Andres replied to my post ;-) I got the same idea then him; and guess what, it's working! So problem is Asterisk 1.8/10 in VM _only_ this provider(s) which are all Dellmont services. Can someone confirm the problem? Question is now, who is faulty? Should I open a bug? Thanks for your time and support. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.4.0 GotoIf to label problem when DUNDi active
I have a hotdesking environment at my main office, and up until today, the
GotoIf that jumps straight to voicemail if a user isn't log in was working just
fine by label. Today, I deployed DUNDi to a satellite office, and now the
GotoIf isn't jumping to the right place. If I replace the label with a
priority number, it jumps correctly. Alternatively, if I disable the switch
statement for DUNDi, it jumps correctly. But with the DUNDi switch in service
and the named label to jump to, it gives me this error:
[May 30 13:57:24] WARNING[6654]: pbx.c:10747 pbx_parseable_goto: Priority
'not_logged_in' must be a number 0, or valid label
Dialplan snippets as follows:
[hotdesk] ;phones dial here
include = hotdesk_outbound
[hotdesk_outbound]
exten = _X.,1,NoOp()
same = n,Set(LOCATION=${CUT(CHANNEL,/,2)})
same = n,Set(LOCATION=${CUT(LOCATION,-,1)})
same = n,Set(WHO=${HOTDESK_PHONE_STATUS(${LOCATION})})
same = n,GotoIf($[${ISNULL(${WHO})}]?internal,${EXTEN},1)
same = n,Set(${WHO}_CID_NAME=${HOTDESK_INFO(cid_name,${WHO})})
same = n,Set(${WHO}_CID_NUMBER=${HOTDESK_INFO(cid_number,${WHO})})
same = n,Set(${WHO}_CONTEXT=${HOTDESK_INFO(defaultcontext,${WHO})})
same = n,Set(${WHO}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${WHO})})
same = n,Set(GROUP(activecallers)=${WHO})
same = n,NoOp(Who: ${WHO} Calls: ${GROUP_COUNT(${WHO}@activecallers)})
same = n,Set(DEVICE_STATE(Custom:${WHO})=INUSE)
same = n,Set(CALLERID(name)=${${WHO}_CID_NAME})
same = n,Set(CALLERID(num)=${${WHO}_CID_NUMBER})
same = n,Goto(${${WHO}_CONTEXT},${EXTEN},1)
[outbound-context]
include = internal-privledged
[internal-privledged]
include = internal
switch = DUNDi/peer
[internal]
exten = _3XX,1,NoOp()
same = n,Set(E=${EXTEN})
same = n,Set(${E}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${E})})
same = n,Set(USER_LOCATION=${HOTDESK_USER_STATUS(${E})})
same = n,GotoIf($[${ODBCROWS} 1]?not_logged_in)
same = n,Dial(SIP/${USER_LOCATION},20,wWU(answered^${E}))
same =
n,ExecIf(${ISNULL(${E})}?NoOp(${HOTDESK_INFO(location,${E})}):ExecIf($[${GROUP_COUNT(${E}@activecalls)}1]?Set(DEVICE_STATE(Custom:${E})=INUSE):Set(DEVICE_STATE(Custom:${E})=NOT_INUSE)))
same = n,Set(GROUP(activecalls)=${NULL})
same = n,Voicemail(${E}@${${E}_VMCONTEXT},b)
same = n,Hangup()
same = n(not_logged_in),Set(LOGGED_OFF=1)
same = n,Voicemail(${E}@${${E}_VMCONTEXT},u)
same = n,Hangup()
Any suggestions on other things to try? Or is this a bug I should file?
Thank you,
Noah Engelberth
MetaLINK Technologies
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Re: [asterisk-users] Fax Server for Asterisk
I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable at all; sometimes the fax reach the destination, sometimes not, and even worse, asterisk got froozen...(here using analog lines over Sangoma B600 and Digium TDM400P, same behavior with both. Other history with same asterisk version but E1 lines, it was PERFECT. That's why i ask for analog lines, since not all customers has E1. Any recommendation/restriction when using hylafax + Asterisk + iaxmodem ? BR El 29/05/2012 22:29, Carlos Alvarez car...@televolve.com escribió: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma Card Issue
Has anyone experienced an issue with Sangoma analog cards where the card suddenly stops working? Trying to dial out shows the channel as busy, even though there is no active call on that port? This happened to us often when we used Digium cards (in fact this issue is why we stopped using Digium). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card Issue
Post more regarding your hardware and software configuration. Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 On 5/30/2012 2:34 PM, Eric Wieling wrote: Has anyone experienced an issue with Sangoma analog cards where the card suddenly stops working? Trying to dial out shows the channel as busy, even though there is no active call on that port? This happened to us often when we used Digium cards (in fact this issue is why we stopped using Digium). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Server for Asterisk
Just to clarify, i were using fax machines connected to fxs ports El 30/05/2012 20:31, Danny Dias ing.diasda...@gmail.com escribió: I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable at all; sometimes the fax reach the destination, sometimes not, and even worse, asterisk got froozen...(here using analog lines over Sangoma B600 and Digium TDM400P, same behavior with both. Other history with same asterisk version but E1 lines, it was PERFECT. That's why i ask for analog lines, since not all customers has E1. Any recommendation/restriction when using hylafax + Asterisk + iaxmodem ? BR El 29/05/2012 22:29, Carlos Alvarez car...@televolve.com escribió: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.12.2 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.12.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.12.2 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: * --- Resolve crash in subscribing for MWI notifications (Closes issue ASTERISK-19827. Reported by B. R) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.2 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.4.2 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.4.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.4.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Resolve crash in subscribing for MWI notifications (Closes issue ASTERISK-19827. Reported by B. R) * --- Fix crash in ConfBridge when user announcement is played for more than 2 users (Closes issue ASTERISK-19899. Reported by Florian Gilcher) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.2 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card Issue
On 5/30/2012 2:34 PM, Eric Wieling wrote: Has anyone experienced an issue with Sangoma analog cards where the card suddenly stops working? Trying to dial out shows the channel as busy, even though there is no active call on that port? This happened to us often when we used Digium cards (in fact this issue is why we stopped using Digium). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have seen this happen when you have a remora card and did not connect the molex power adapter, even if you only use FXO ports. Sometimes it works and sometimes it does not. Its better to always connect it to power. -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Server for Asterisk
I had Hylafax sending 1000s of faxes a day twice a week connected to analogue lines using asterisk and iaxmodem for about 4 years. People don't use fax much anymore though No problems whatsoever Cheers Duncan On 31/05/2012, at 6:49 AM, Danny Dias wrote: Just to clarify, i were using fax machines connected to fxs ports El 30/05/2012 20:31, Danny Dias ing.diasda...@gmail.com escribió: I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable at all; sometimes the fax reach the destination, sometimes not, and even worse, asterisk got froozen...(here using analog lines over Sangoma B600 and Digium TDM400P, same behavior with both. Other history with same asterisk version but E1 lines, it was PERFECT. That's why i ask for analog lines, since not all customers has E1. Any recommendation/restriction when using hylafax + Asterisk + iaxmodem ? BR El 29/05/2012 22:29, Carlos Alvarez car...@televolve.com escribió: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card Issue
On Wed, May 30, 2012 at 02:34:55PM -0400, Eric Wieling wrote: Has anyone experienced an issue with Sangoma analog cards where the card suddenly stops working? Trying to dial out shows the channel as busy, even though there is no active call on that port? This happened to us often when we used Digium cards (in fact this issue is why we stopped using Digium). Odd, I'm not aware of any current issues with Digium's cards or drivers which would leave it in an alarm state that wasn't attributable to an intermittent cabling issue. I've seen a loss of voltage between the tip and ring (assuming you're talking about FXO ports) due to cabling issues that would leave the card in alarm. If it's something marginal you could sometimes play with the battery threshold. I had seen some issues with the newer analog cards and stuck alarm states that were addressed in r7517 voicebus: send 'idle' buffers when the transmit descriptor underruns [1] which was first released in dahdi-linux 2.3.0. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=7517 If you still have the cards with which this problem occurred often, are able to reproduce it with the dahdi-linux 2.6.1, I would be interested in determining what the problem is. If you are interested and able as well, shoot me an email off-list. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card Issue
I switched to using Sangoma cards around 2003 or 2004 at my previous employer. My current employer was using Digium cards up to 2008 and had the issue. We have no interest in switching back to Digium. I did not mean to imply that we had issues with current Digium cards, but re-reading my original message, I can see how I might have given that impression. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Wednesday, May 30, 2012 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma Card Issue On Wed, May 30, 2012 at 02:34:55PM -0400, Eric Wieling wrote: Has anyone experienced an issue with Sangoma analog cards where the card suddenly stops working? Trying to dial out shows the channel as busy, even though there is no active call on that port? This happened to us often when we used Digium cards (in fact this issue is why we stopped using Digium). Odd, I'm not aware of any current issues with Digium's cards or drivers which would leave it in an alarm state that wasn't attributable to an intermittent cabling issue. I've seen a loss of voltage between the tip and ring (assuming you're talking about FXO ports) due to cabling issues that would leave the card in alarm. If it's something marginal you could sometimes play with the battery threshold. I had seen some issues with the newer analog cards and stuck alarm states that were addressed in r7517 voicebus: send 'idle' buffers when the transmit descriptor underruns [1] which was first released in dahdi-linux 2.3.0. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=7517 If you still have the cards with which this problem occurred often, are able to reproduce it with the dahdi-linux 2.6.1, I would be interested in determining what the problem is. If you are interested and able as well, shoot me an email off-list. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card Issue
On Wed, May 30, 2012 at 04:35:25PM -0400, Eric Wieling wrote: I did not mean to imply that we had issues with current Digium cards, but re-reading my original message, I can see how I might have given that impression. Ahh, ok. Then that makes sense. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX ATA can't register
I have an ATCOM ATA that is trying to connect to an asterisk server using IAX. The ATA and Asterisk are on the same subnet, not firewall/nat etc. Below is a a log excerpt, showing the REGREQ received, and then Asterisk goes on to send lots of REGAUTH...and this continues for a while, but the ATA is never registered (iax2 show peers shows not registered). Any help would be appreciated. It sure LOOKS like a lot of TX for very few RX frames...so my first guess was network related but I'm not making any progress with that theory Thanks Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 21003ms SCall: 10027 DCall: 18442 [192.168.67.20:4569] Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 21006ms SCall: 14940 DCall: 18442 [192.168.67.20:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 18443 DCall: 0 [192.168.67.20:4569] USERNAME: ALARM-ATA REFRESH : 60 Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00014ms SCall: 02267 DCall: 18443 [192.168.67.20:4569] AUTHMETHODS : 2 CHALLENGE : 121149566 USERNAME: ALARM-ATA Tx-Frame Retry[002] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10013ms SCall: 02267 DCall: 18443 [192.168.67.20:4569] Tx-Frame Retry[001] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 20013ms SCall: 02267 DCall: 18443 [192.168.67.20:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 21016ms SCall: 10572 DCall: 18443 [192.168.67.20:4569] Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00015ms SCall: 14112 DCall: 18443 [192.168.67.20:4569] AUTHMETHODS : 2 CHALLENGE : 530555480 USERNAME: ALARM-ATA Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 21015ms SCall: 12659 DCall: 18443 [192.168.67.20:4569] Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10014ms SCall: 14112 DCall: 18443 [192.168.67.20:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 20016ms SCall: 00480 DCall: 18443 [192.168.67.20:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 2ms SCall: 05489 DCall: 18443 [192.168.67.20:4569] AUTHMETHODS : 2 CHALLENGE : 399007934 USERNAME: ALARM-ATA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Server for Asterisk
- Original Message - I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable at all; sometimes the fax reach the destination, sometimes not, and even worse, asterisk got froozen...(here using analog lines over Sangoma B600 and Digium TDM400P, same behavior with both. Other history with same asterisk version but E1 lines, it was PERFECT. That's why i ask for analog lines, since not all customers has E1. Any recommendation/restriction when using hylafax + Asterisk + iaxmodem ? BR If Asterisk was freezing up, that would seem to indicate a problem with Asterisk, not the Hylafax/IAXmodem components. Of course, details would be needed to determine why that was the case. Regardless, without lockups of Asterisk, reliability of fax is very dependent on timing and audio quality. Again, details would be needed to further investigate why you had high failure metrics(specifically your fax session logs from /var/spool/hylafax/log). In general, Hylafax+[1] and IAXmodem is the most rock solid stable fax solution available, as long as you can get past the initial learning curve. There is a reason why IAXmodem has not had a release in forever as the 1.2.0 release is rock solid stable. Hylafax+ continues to be developed with regular releases, the feature set and functionality are second to none with hooks for almost any imaginable configuration, and the support via the mailing lists or available contractors can't be beat. /soapbox If you have specifics about your problems with Hylafax and IAXmodem, I'd love to hear about them to help diagnose, if it is postmortem. --Tim [1] There *IS* a difference between Hylafax (hylafax.org) and Hylafax+ (hylafax.sourceforge.net). Please see here: http://hylafax.sourceforge.net/about.php -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 10.4.0 GotoIf to label problem when DUNDi active
Hi,
You might have already tried but can you try reducing the label name and
exclude the underscore in it !
Regards,
Sammy
On Wed, May 30, 2012 at 11:02 PM, Noah Engelberth
n...@directlinkcomputers.com wrote:
I have a hotdesking environment at my main office, and up until today,
the GotoIf that jumps straight to voicemail if a user isn’t log in was
working just fine by label. Today, I deployed DUNDi to a satellite office,
and now the GotoIf isn’t jumping to the right place. If I replace the
label with a priority number, it jumps correctly. Alternatively, if I
disable the switch statement for DUNDi, it jumps correctly. But with the
DUNDi switch in service and the named label to jump to, it gives me this
error:
** **
[May 30 13:57:24] WARNING[6654]: pbx.c:10747 pbx_parseable_goto: Priority
'not_logged_in' must be a number 0, or valid label
** **
Dialplan snippets as follows:
[hotdesk] ;phones dial here
include = hotdesk_outbound
** **
[hotdesk_outbound]
exten = _X.,1,NoOp()
same = n,Set(LOCATION=${CUT(CHANNEL,/,2)})
same = n,Set(LOCATION=${CUT(LOCATION,-,1)})
same = n,Set(WHO=${HOTDESK_PHONE_STATUS(${LOCATION})})
same = n,GotoIf($[${ISNULL(${WHO})}]?internal,${EXTEN},1)
same = n,Set(${WHO}_CID_NAME=${HOTDESK_INFO(cid_name,${WHO})})
same = n,Set(${WHO}_CID_NUMBER=${HOTDESK_INFO(cid_number,${WHO})})
same = n,Set(${WHO}_CONTEXT=${HOTDESK_INFO(defaultcontext,${WHO})})
same = n,Set(${WHO}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${WHO})})
same = n,Set(GROUP(activecallers)=${WHO})
same = n,NoOp(Who: ${WHO} Calls: ${GROUP_COUNT(${WHO}@activecallers)})***
*
same = n,Set(DEVICE_STATE(Custom:${WHO})=INUSE)
same = n,Set(CALLERID(name)=${${WHO}_CID_NAME})
same = n,Set(CALLERID(num)=${${WHO}_CID_NUMBER})
same = n,Goto(${${WHO}_CONTEXT},${EXTEN},1)
** **
[outbound-context]
include = internal-privledged
** **
[internal-privledged]
include = internal
switch = DUNDi/peer
** **
[internal]
exten = _3XX,1,NoOp()
same = n,Set(E=${EXTEN})
same = n,Set(${E}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${E})})
same = n,Set(USER_LOCATION=${HOTDESK_USER_STATUS(${E})})
same = n,GotoIf($[${ODBCROWS} 1]?not_logged_in)
same = n,Dial(SIP/${USER_LOCATION},20,wWU(answered^${E}))
same =
n,ExecIf(${ISNULL(${E})}?NoOp(${HOTDESK_INFO(location,${E})}):ExecIf($[${GROUP_COUNT(${E}@activecalls
)}1]?Set(DEVICE_STATE(Custom:${E})=INUSE):Set(DEVICE_STATE(Custom:${E})=NOT_INUSE)))
same = n,Set(GROUP(activecalls)=${NULL})
same = n,Voicemail(${E}@${${E}_VMCONTEXT},b)
same = n,Hangup()
same = n(not_logged_in),Set(LOGGED_OFF=1)
same = n,Voicemail(${E}@${${E}_VMCONTEXT},u)
same = n,Hangup()
** **
Any suggestions on other things to try? Or is this a bug I should file?**
**
** **
** **
Thank you,
** **
Noah Engelberth
MetaLINK Technologies
** **
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