[asterisk-users] Spam and that recent 'attack' ...
Here's an intersting thing - in my spam folder today in popped an email from a Chinese manufacturing company trying to sell me the usual VoIP stuff - phones, etc. (From com-vox.com FWIW). However as well as targetting the email address on my web site, they also targetted a made-up, but plausable, email address at the host that was attacked last weekend - (syst...@drogon.net and syst...@x.drogon.net) and that host has no public mentions anywhere that I'm aware of (Although once upon a time I did publish a SIP URI to it, but that was removed well over a year ago). Also that host doesn't run email and has no MX records pointing to it. So it's probably not related, but just a passing curiosity... Gordon -- _ -- 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] Transfer_CONTEXT behaviour
On 16 April 2010 02:53, Jared Smith jsm...@digium.com wrote: You'll need to play around with variable inheritance to get it set right. If you define a variable with a single underscore (_TRANSFER_CONTEXT in my example), it'll get inherited by the next spawned channel, but go no further. If you define a variable with two underscores (say, __TRANSFER_CONTEXT), then it will get inherited by the next spawned channel, and any channels spawned by that channel, and so forth. Obviously defining it without any underscores at all means it won't get inherited by spawned channels. Excellent - Thanks Jared. This will save a lot of effort :) Regards, Steve -- _ -- 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] Friday April 16 @12 Noon EDT - Tim's Excellent Island Telephony Adventure, AstriEurop, and more EC2 rant
Hey, This week I return from AstriEurop and I'm lucky because I left yesterday, many of my Asterisk friends are stranded because of the air and rail strikes that got worse today. I will mention a few things I think are interesting from my meetings there and hope a few others will make it and share their experiences. in particular, Mark Spencer's keynote held a surprise or two. Also, we will be talking about the situation of EC2 attacks. I am adamant about getting Amazon in on this discussion. We'll compare notes on the conference later today about this more than irritating situation. Finally, the featured guest Tim Panton from PhoneFromHere.com and hopefully David Burgess from Open BTS will share with us some of their travails on the distant island where the .nu domain resides. Something for everyone! Rants about EC2 attacks, detailed evaluation of the swag from AstriEurop (and rants about the wifi) and then some actual VoIP discussion about a very interesting technology and how it's helping people in remote areas in a big way. sip:200...@login.zipdx.com starting from 11:45 AM EDT (5:45PM Europe) in g722 wideband or g711 skype:vuc.me - Yay Skype for Asterisk and thanks to Tim Panton and Digium for all the fish IRC: http://vuc.me/irc or #vuc -- _ -- 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] SS7 over an FXO interface
Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 - connected to - FXO interface in PBX2 = used to transport ss7 signaling. - FXS interface in PBX2 - connected to - FXO interface in PBX1 = used to transport voice between the two PBXs. This connection can be replaced by a simple SIP trunk. Is this scenario possible with libss7 and asterisk. If yes, please give some instructions and tips. Thanks. -- _ -- 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] SIP devide call-forward behaviour and CDRs
On 15 April 2010 18:11, Steve Davies davies...@gmail.com wrote: Hi, I am migrating some billing code from 1.2 to 1.6 cdr output. Mostly this is not too bad, but I have a scenario where some data appears to be lost Call from SIP/100 to SIP/200, but the SIP/200 device is programmed to send a redirect to extension 1234. chan_sip creates a Local/1...@context call, which has its own CDR. In 1.2, the CDR records look something like: 1) channel=SIP/100 dstchannel=Local/1...@context-deadbeef,1 2) channel=Local/1...@context-deadbeef,2 dstchannel=DAHDI/1-1 This allowed the 2 Local/... values to be matched and the 2 legs of the call can be correctly billed. In 1.6, the CDR data is different: 1) channel=SIP/100 dstchannel=DAHDI/1-1 2) channel=Local/1...@context-deadbeef;2 dstchannel=DAHDI/1-1 and we lose the ;1 half of the local channel pair and the data is virtually useless :( Is this an intentional change, and can I put it back somehow? I am wondering if I need to add /n to the channel string that is built by chan_sip so that the bridge does not destroy the local channel? Thanks, Steve Following up to my own post... Sorry. I also notice that if I Dial(Local/1...@context) in the dialplan, I still get the old style (correct) data whether I use /n on the local channel or not. The only difference I can see is the SIP call_forward request to emulate a SIP hairpin. I thought this call forward mechanism basically destroyed the original call and replaced it - I am not sure how or why that would affect the cdr of the original inbound call? Regards, Steve -- _ -- 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] SS7 over an FXO interface
On 4/16/10 3:15 PM, mosbah abdelkader wrote: Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 - connected to - FXO interface in PBX2 = used to transport ss7 signaling. - FXS interface in PBX2 - connected to - FXO interface in PBX1 = used to transport voice between the two PBXs. This connection can be replaced by a simple SIP trunk. Is this scenario possible with libss7 and asterisk. If yes, please give some instructions and tips. Thanks. NO, you can't pass signaling over an analog FXO. You can pass SS7 over Ethernet or E1/T1 links, least to mention. -- _ -- 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] Delay the HungUp
Hi! exten = h,1,NoOp(Start) exten = h,n,Wait(5) exten = h,n,NoOp(End) exten = h,n,Hangup() * Wait() does not work in the context of the h extension (I am curious as to the reasons for that as well) * Hangup() makes no sense in the context of the h extension and should not be used; you are _already_ in the process of hanging up Philipp -- _ -- 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] Delay the HungUp
I think you should try to use option 'g' in the Dial command and put Wait after that, not in the h extension. But what exactly you are trying to achievee here? Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-16 8:03 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! exten = h,1,NoOp(Start) exten = h,n,Wait(5) exten = h,n,NoOp(End) exten = h,n,Hangup() ... * Wait() does not work in the context of the h extension (I am curious as to the reasons for that as well) * Hangup() makes no sense in the context of the h extension and should not be used; you are _already_ in the process of hanging up Philipp -- _ -- Bandwidth and Colocat... -- _ -- 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] Delay the HungUp
The h extension does not get executed until the hang up has occurred. The use of this is to do something after the hang up. Like cleaning up database entires or sending user events or the like. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 16, 2010, at 4:58 AM, Philipp von Klitzing wrote: Hi! exten = h,1,NoOp(Start) exten = h,n,Wait(5) exten = h,n,NoOp(End) exten = h,n,Hangup() * Wait() does not work in the context of the h extension (I am curious as to the reasons for that as well) * Hangup() makes no sense in the context of the h extension and should not be used; you are _already_ in the process of hanging up Philipp -- _ -- 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] IAX connection slow between 1.4 and 1.6 dists
Hi gang, I'm running 1.4.26.2 and 1.4.30 on my two real asterisk boxes. I just installed 1.6.2.6 on a Suse VM and the phone to asterisk connection works fine. When I try to do an IAX connection from the 1.4 boxes to the 1.6 boxes, the audio runs like molasses. Instead of welcome to blah blah blah, I get wel --- Come to bla #$#$$$ blah ##$$#$$ blah $#$#$. Any pointers? Thanks Danny Nicholas -- _ -- 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] Delay the HungUp
Hi, Thanks for your answer. We need to delay the HungUp because some calls that we dial are so short (3 or 4 seconds) and our provider requests 8 seconds. That is the reason. I will try with g option. Thanks again Zeeshan Zakaria wrote: I think you should try to use option 'g' in the Dial command and put Wait after that, not in the h extension. But what exactly you are trying to achievee here? Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-16 8:03 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de mailto:klitz...@pool.informatik.rwth-aachen.de wrote: Hi! exten = h,1,NoOp(Start) exten = h,n,Wait(5) exten = h,n,NoOp(End) exten = h,n,Hangup() ... * Wait() does not work in the context of the h extension (I am curious as to the reasons for that as well) * Hangup() makes no sense in the context of the h extension and should not be used; you are _already_ in the process of hanging up Philipp -- _ -- Bandwidth and Colocat... -- _ -- 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] Delay the HungUp
On 16 Apr 2010, at 14:39, cbulist wrote: We need to delay the HungUp because some calls that we dial are so short (3 or 4 seconds) and our provider requests 8 seconds. That is the reason Sounds like you need a decent provider ;) S -- _ -- 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] Recording music in Queue
I know that this is a feature but I would like to have the hold music recorded while a person is on hold. So I know the agent put them on hold and not just muted. I have monitor-join=yes monitor-format=wav in my queues.conf any ideas? Per http://www.asteriskguru.com/tutorials/queues_conf.html The best part is no recording will be initiated while the people are listening to music on hold Jason Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. -- _ -- 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] Recording music in Queue
Instead of putting them on hold, transfer them to a recorded muted conference; the agent would pick them up by joining and un-muting the conference. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Walker Sent: Friday, April 16, 2010 9:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Recording music in Queue I know that this is a feature but I would like to have the hold music recorded while a person is on hold. So I know the agent put them on hold and not just muted. I have monitor-join=yes monitor-format=wav in my queues.conf any ideas? Per http://www.asteriskguru.com/tutorials/queues_conf.html The best part is no recording will be initiated while the people are listening to music on hold Jason Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. -- _ -- 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] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)
Danny Dias wrote: We are having MANY but MANY problems configuring an analog fax machine to work properly on Asterisk, the first thing we do was to plug in the Fax analog machine to the FXS port of the Digium TDM410P, we set echocancel=no on zapata.conf and also faxdetect=yes on general section, but our Asterisk CRASH every time we try to send/receive fax! We are using Asterisk 1.4.21 and Zaptel 1.4.11; the card does not shows any interrupt in /proc/interrupts We also tried with a Sangoma B600 on this machine and the same result! Then we tried with the sangoma and digium card on another asterisk box, with Asterisk 1.4.30 and DAHDI 2.2.x and the fax was not reliable 100% but at least Asterisk vener went down So you've confirmed that a later release of Asterisk solves the problem you are trying to resolve, but your customer won't let you use that version. What do you expect people here to be able to do to assist you? Either you upgrade to the version that has the problem resolved, or you figure out how to backport the relevant changes into the version you are running... but that seems to also violate your customer's restriction that you can't change Asterisk versions. -- 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 -- 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] How can I record the conversations in a conference call?
Please note: A Zaptel timer must be present for conferencing to work!, but if the user does not have ZAP/DAHDI hardware, he can use ZAP/DAHDI DUMMY Actually, my understanding is that this is incorrect. The conference must contain ZAP/DAHDI callers. A dummy won't do. The reason is that the ZAP/DAHDI driver mixes the audio in the driver and when this is not available it falls back to mixing within MeetMe. But in such case, you can neither record the conference nor run an AGI in the background. See: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe. This is from 2004, maybe it changed by now. Luki -- _ -- 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] incoming ghost call
Hello asterisk users... We are having a little problem in our installation, we are using Asterisk 1.4.21.2 and zaptel 1.4.11 with a Digium TDM410P (3FXO + 1FXS), the problem is that when we disconnect the line from any of the fxo ports, we receive an incoming ghost call (using zap/x channel) it rings on the phone but we cant hear nothing...it's always doing the same everytime we disconnect the lines from the fxo We tried with a Sangoma card, and the problem went away, but we must use this digium card, we've tried with answer/hangup on polarityswitch with all the options, and we cant make this work ok, what should we do? Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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] Recording music in Queue
On Fri, 2010-04-16 at 09:00 -0500, Jason Walker wrote: I know that this is a “feature” but I would like to have the hold music recorded while a person is on hold. So I know the agent put them on hold and not just muted. I have monitor-join=yes monitor-format=wav in my queues.conf Once the call gets to an agent it will be recorded, even the music if the user is on hold. What you quoted is that the recording will not start UNTIL the agent has answered the call from the queue. -- 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 -- 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] On CLI SIP don't appear
Hi, Anyone know why sometimes on CLI disappear parameters as sip, stop,...? Thank you very much by reply. Renato -- _ -- 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] How can I record the conversations in a conference call?
On Fri, 2010-04-16 at 08:38 -0700, Luki wrote: Please note: A Zaptel timer must be present for conferencing to work!, but if the user does not have ZAP/DAHDI hardware, he can use ZAP/DAHDI DUMMY Actually, my understanding is that this is incorrect. The conference must contain ZAP/DAHDI callers. A dummy won't do. The reason is that the ZAP/DAHDI driver mixes the audio in the driver and when this is not available it falls back to mixing within MeetMe. But in such case, you can neither record the conference nor run an AGI in the background. See: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe. This is from 2004, maybe it changed by now. Luki I use Meetme all the time in servers that just use dahdi_dummy. I would say that in the past no one would recommend having more than 10 users in a conference is you did not have a hardware clock but that has changed. With newer kernels and Asterisk versions I have been able to get over 50 people in a single Meetme room without any glitches. -- 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 -- 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] How can I record the conversations in a conference call?
On Fri, 16 Apr 2010, Carlos Chavez wrote: On Fri, 2010-04-16 at 08:38 -0700, Luki wrote: Please note: A Zaptel timer must be present for conferencing to work!, but if the user does not have ZAP/DAHDI hardware, he can use ZAP/DAHDI DUMMY Actually, my understanding is that this is incorrect. The conference must contain ZAP/DAHDI callers. A dummy won't do. The reason is that the ZAP/DAHDI driver mixes the audio in the driver and when this is not available it falls back to mixing within MeetMe. But in such case, you can neither record the conference nor run an AGI in the background. See: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe. This is from 2004, maybe it changed by now. Luki I use Meetme all the time in servers that just use dahdi_dummy. I would say that in the past no one would recommend having more than 10 users in a conference is you did not have a hardware clock but that has changed. With newer kernels and Asterisk versions I have been able to get over 50 people in a single Meetme room without any glitches. I also run all SIP conferences with dahdi_dummy and have recorded them. As old as 1.4.22.1 seems to work fine. Using options riM and setting MEETME_RECORDINGFILE beforehand. j -- _ -- 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] IAX connection slow between 1.4 and 1.6 dists
- Original Message - Hi gang, I’m running 1.4.26.2 and 1.4.30 on my two “real” asterisk boxes. I just installed 1.6.2.6 on a Suse VM and the phone to asterisk connection works fine. When I try to do an IAX connection from the 1.4 boxes to the 1.6 boxes, the audio runs like molasses. Instead of “welcome to blah blah blah”, I get “wel --- Come to bla #$#$$$ blah ##$$#$$ blah $#$#$”. Any pointers? It likely has nothing to do with the versions of Asterisk you're running and more to do with the fact that one 'box' is inside of a VM which can give poor performance/timing. Are you by any chance using IAX 'trunk=yes'? This requires a proper timing source... Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connection slow between 1.4 and 1.6 dists
Good Idea, Tim - did have trunk=yes on both boxes. Changed to trunk=no and retried - a little better molasses, but still molasses. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, April 16, 2010 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX connection slow between 1.4 and 1.6 dists - Original Message - Hi gang, I'm running 1.4.26.2 and 1.4.30 on my two real asterisk boxes. I just installed 1.6.2.6 on a Suse VM and the phone to asterisk connection works fine. When I try to do an IAX connection from the 1.4 boxes to the 1.6 boxes, the audio runs like molasses. Instead of welcome to blah blah blah, I get wel --- Come to bla #$#$$$ blah ##$$#$$ blah $#$#$. Any pointers? It likely has nothing to do with the versions of Asterisk you're running and more to do with the fact that one 'box' is inside of a VM which can give poor performance/timing. Are you by any chance using IAX 'trunk=yes'? This requires a proper timing source... Tim -- _ -- 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] Need help with a pika warp asterisk appliance problem.
Timothy, The Warp analog modules version A.1 are not compatible with the PADS 2.x software base. Replacing the modules to A.3 or newer and upgrading the software to PADS 2.1 or newer should resolve these issues. Best Regards, Rod Boileau Manager, Customer Care PIKA Technologies -- _ -- 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] IAX connection slow between 1.4 and 1.6 dists
I run a virtual 64 bit Cent OS 5.4 system under VMWare's Fusion 3 with asterisk 1.4 IAX trunked from my house to my office 600 miles away. I do not do more than a couple concurrent calls but I have no problem recording calls, having meetme conferences, playing sound. I have even connected a Xorcom unit and used it to connect to an analog line and phone without problem. This all on a two or three year old 2.16GHz Core Duo 2GB memory MacBook 13. For testing this has served me well. I do feel that virtual systems are viable for many tasks. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 16, 2010, at 10:30 AM, Danny Nicholas wrote: Good Idea, Tim - did have trunk=yes on both boxes. Changed to trunk=no and retried - a little better molasses, but still molasses. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, April 16, 2010 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX connection slow between 1.4 and 1.6 dists - Original Message - Hi gang, I'm running 1.4.26.2 and 1.4.30 on my two real asterisk boxes. I just installed 1.6.2.6 on a Suse VM and the phone to asterisk connection works fine. When I try to do an IAX connection from the 1.4 boxes to the 1.6 boxes, the audio runs like molasses. Instead of welcome to blah blah blah, I get wel --- Come to bla #$#$$$ blah ##$$#$$ blah $#$#$. Any pointers? It likely has nothing to do with the versions of Asterisk you're running and more to do with the fact that one 'box' is inside of a VM which can give poor performance/timing. Are you by any chance using IAX 'trunk=yes'? This requires a proper timing source... Tim -- _ -- 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 -- _ -- 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] IAX connection slow between 1.4 and 1.6 dists
Thanks for the good word, Jim. Apparently it is some kind on 1.6 issue; I installed 1.4.30 and voila - perfect sound!! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Friday, April 16, 2010 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX connection slow between 1.4 and 1.6 dists I run a virtual 64 bit Cent OS 5.4 system under VMWare's Fusion 3 with asterisk 1.4 IAX trunked from my house to my office 600 miles away. I do not do more than a couple concurrent calls but I have no problem recording calls, having meetme conferences, playing sound. I have even connected a Xorcom unit and used it to connect to an analog line and phone without problem. This all on a two or three year old 2.16GHz Core Duo 2GB memory MacBook 13. For testing this has served me well. I do feel that virtual systems are viable for many tasks. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 16, 2010, at 10:30 AM, Danny Nicholas wrote: Good Idea, Tim - did have trunk=yes on both boxes. Changed to trunk=no and retried - a little better molasses, but still molasses. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Friday, April 16, 2010 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX connection slow between 1.4 and 1.6 dists - Original Message - Hi gang, I'm running 1.4.26.2 and 1.4.30 on my two real asterisk boxes. I just installed 1.6.2.6 on a Suse VM and the phone to asterisk connection works fine. When I try to do an IAX connection from the 1.4 boxes to the 1.6 boxes, the audio runs like molasses. Instead of welcome to blah blah blah, I get wel --- Come to bla #$#$$$ blah ##$$#$$ blah $#$#$. Any pointers? It likely has nothing to do with the versions of Asterisk you're running and more to do with the fact that one 'box' is inside of a VM which can give poor performance/timing. Are you by any chance using IAX 'trunk=yes'? This requires a proper timing source... Tim -- _ -- 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 -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Hello! I have developed an IVR using AGI and so far it works great. I'm using Cepstral voices, but now want to use the voices from AT T that are on a Windows server to be heard best. With cepstral what I do is to generate audio files from shipping and this text I reproduce this method it has worked very well. Now, try to do the same by creating the audio file in windows with the voices of AT T, the problem is that there is no way to synchronize the generation of the audio file and step Asterisk to be played, so it occurred to me to use FastAGI to generate all Windows and play in the same window the audio file generated. We buy Linux licenses for the voices but they are very expensive and already bought windows for another project. How do you think would be the best option? If you have another idea, please Tell me because I'm getting crazy with this and can not solve. TIA Edwin _ -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Why don't you use sox to transform the windows audio file into the asterisk format - I do this with pretty good results. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Edwin Quijada Sent: Friday, April 16, 2010 3:59 PM To: Asterisk Asterisk Subject: [asterisk-users] AGI, FASTAGI or Windows Voice Server Hello! I have developed an IVR using AGI and so far it works great. I'm using Cepstral voices, but now want to use the voices from AT T that are on a Windows server to be heard best. With cepstral what I do is to generate audio files from shipping and this text I reproduce this method it has worked very well. Now, try to do the same by creating the audio file in windows with the voices of AT T, the problem is that there is no way to synchronize the generation of the audio file and step Asterisk to be played, so it occurred to me to use FastAGI to generate all Windows and play in the same window the audio file generated. We buy Linux licenses for the voices but they are very expensive and already bought windows for another project. How do you think would be the best option? If you have another idea, please Tell me because I'm getting crazy with this and can not solve. TIA Edwin _ -- _ -- 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] Testing a sip call through Asterisk?
I'm looking to find a test tool that will register with our Asterisk (Trixbox) server here at work and place an outgoing call via our main SIP trunk (BroadVoice) to confirm that things are working. I've looked around but I can't seem to find any tools that will do what I'm looking for. I can't just monitor the status of the trunk inside Asterisk, as this is the normal status: asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status BroadVoice/425256 147.135.32.221 N 5060 Unmonitored ... 37 sip peers [Monitored: 5 online, 31 offline Unmonitored: 1 online, 0 offline] asterisk*CLI Alternatively, any suggestions as to how I can change the trunk configuration so that it is monitored would be appreciated. The peer config is set as: allow=ulaw disallow=all canreinvite=no context=from-trunk dtmf=inband dtmfmode=inband fromdomain=sip.broadvoice.com fromuser=425256 host=sip.broadvoice.com insecure=very nat=yes secret=XX type=peer username=425256 Any assistance would be appreciated. I'd rather know when things fail via an automated system rather than learning it's down from the users. -- Nathan Clemons -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Why don’t you use sox to transform the windows audio file into the asterisk format – I do this with pretty good results. I did. But my problem is not conversion my problem is that I dont know how play the file from windows server or copy this to asterisk without my AGI continue and desyncronyze it. Can you explain me exactly what did you do /? Do you have something like this using AGI ? I use sox with good results too in windows. The problem is when create the file and convert it , how send to asterisk Edwin Jaws _ -- _ -- 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] Testing a sip call through Asterisk?
On 04/16/2010 03:39 PM, Nathan Clemons wrote: I'm looking to find a test tool that will register with our Asterisk (Trixbox) server here at work and place an outgoing call via our main SIP trunk (BroadVoice) to confirm that things are working. I've looked around but I can't seem to find any tools that will do what I'm looking for. I can't just monitor the status of the trunk inside Asterisk, as this is the normal status: asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status BroadVoice/425256 147.135.32.221 N 5060 Unmonitored ... 37 sip peers [Monitored: 5 online, 31 offline Unmonitored: 1 online, 0 offline] asterisk*CLI Alternatively, any suggestions as to how I can change the trunk configuration so that it is monitored would be appreciated. The peer config is set as: allow=ulaw disallow=all canreinvite=no context=from-trunk dtmf=inband dtmfmode=inband fromdomain=sip.broadvoice.com http://sip.broadvoice.com fromuser=425256 host=sip.broadvoice.com http://sip.broadvoice.com insecure=very nat=yes secret=XX type=peer username=425256 Any assistance would be appreciated. I'd rather know when things fail via an automated system rather than learning it's down from the users. -- Nathan Clemons I believe that adding qualify=enter your value in seconds here to your trunk configuration is what you are looking for for the monitoring state. This will send SIP OPTIONS packets to the trunk periodically. See qualify in the sip.conf samples or documentation. From there you can use a monitoring solution to monitor the state of the trunk. Alternatively you can use a OSS tool called SIPp to test SIP devices. See *http://sipp*.sourceforge.net for more information. This is an indispensable tool for SIP and Asterisk troubleshooting. I hope this helps. -- _ -- 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] Testing a sip call through Asterisk?
On Fri, 16 Apr 2010, Nathan Clemons wrote: I'm looking to find a test tool that will register with our Asterisk (Trixbox) server here at work and place an outgoing call via our main SIP trunk (BroadVoice) to confirm that things are working. I've looked around but I can't seem to find any tools that will do what I'm looking for. I can't just monitor the status of the trunk inside Asterisk, as this is the normal status: [snip] just add qualify=yes to your context and it will monitor the RT latency. j -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Why don't you copy the files to your asterisk box and play them from there? -- Sent from my Android device On Apr 16, 2010 5:03 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Why don’t you use sox to transform the windows audio file into the asterisk format – I do this with ... I did. But my problem is not conversion my problem is that I dont know how play the file from windows server or copy this to asterisk without my AGI continue and desyncronyze it. Can you explain me exactly what did you do /? Do you have something like this using AGI ? I use sox with good results too in windows. The problem is when create the file and convert it , how send to asterisk Edwin Jaws -- -- _ -- 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] On CLI SIP don't appear
On 04/16/2010 10:10 AM, Renato bianchini wrote: Hi, Anyone know why sometimes on CLI disappear parameters as sip, stop,...? Thank you very much by reply. Renato AFAIK when a CLI option is not available it means that module isn't loaded. Check the logs to make sure that module was loaded properly. -- _ -- 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] Testing a sip call through Asterisk?
Turning on qualify=yes, or qualify=60, seems to break the BroadVoice connection (it goes from UNKNOWN to UNREACHABLE and calls fail). I'm wondering if they don't support OPTIONS probing or something. -- Nathan Clemons On Fri, Apr 16, 2010 at 3:22 PM, Jeff LaCoursiere j...@jeff.net wrote: On Fri, 16 Apr 2010, Nathan Clemons wrote: I'm looking to find a test tool that will register with our Asterisk (Trixbox) server here at work and place an outgoing call via our main SIP trunk (BroadVoice) to confirm that things are working. I've looked around but I can't seem to find any tools that will do what I'm looking for. I can't just monitor the status of the trunk inside Asterisk, as this is the normal status: [snip] just add qualify=yes to your context and it will monitor the RT latency. j -- _ -- 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] IAX connection slow between 1.4 and 1.6 dists
On 4/16/2010 3:10 PM, Jim Dickenson wrote: I run a virtual 64 bit Cent OS 5.4 system under VMWare's Fusion 3 with asterisk 1.4 IAX trunked from my house to my office 600 miles away. I do not do more than a couple concurrent calls but I have no problem recording calls, having meetme conferences, playing sound. I have even connected a Xorcom unit and used it to connect to an analog line and phone without problem. This all on a two or three year old 2.16GHz Core Duo 2GB memory MacBook 13. For testing this has served me well. I do feel that virtual systems are viable for many tasks. I certainly did not mean to imply that virtualized Asterisk simply does not work. In fact, I run *MANY* instances in various forms of production, development, testing, home use, etc in Vmware ESX, OpenVZ, Xen, and Virtualbox environments. My thoughts were simply that virtualization and Asterisk can sometimes cause complex issues that require some additional attention than a normal bare metal installation. Tim -- _ -- 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] cat /proc/zaptel/*
Quoting Jaap Winius jwin...@umrk.to: Being both impatient and charitable, I'll try answering this myself: ISDN uses LAPD for the D-channel and LAPB for data connections over the B-channels. However, LAPB is irrelevant for Asterisk, because when the B-channels are used for voice they carry no signaling. This is why it is necessary to specify a line code protocol, such as AMI, for the B-channels, and a frame type, typically CCS, for the D-channel. Would that statement be correct? Basically, although the last line is a little muddled. Also, would someone care to elaborate on how the CCS protocol fits into this picture, in particular how it relates to LAPD? LAPD, described in ITU-T recommendations Q.920 and Q.921, is an OSI network layer 2 protocol, while CCS (Common Channel Signaling), which is described by Q.930 (I.450) and Q.931 (I.451), is layer 3. Two things that I found confusing here are: 1.) The documentation that explains the Zaptel span configuration statement (in /etc/zaptel.conf) describes the D-channel signaling type as framing, which I find misleading. IMO signaling would have been more accurate. 2.) CCS is a connection control signaling type. The problem is that there is more than one CCS type, although my impression is that the one used most often for the ISDN D-channel is Q.930/Q.931. The others I've heard of are QSIG CCS (Q.931/Q.933) and SS7 (Q.700-series with many variants). Cheers, Jaap -- _ -- 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] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)
Can't upgrade the version. how about buying a FXS gateway and be done with the issue. Go to ebay and search for AudioCodes. You can get 1 FXS port gateway for around $30 to 2 FXS at $85. Probably the best bet is to convince the customer to upgrade Asterisk. From: Danny Dias ing.diasda...@gmail.com To: asterisk-users@lists.digium.com Sent: Thu, April 15, 2010 10:16:26 PM Subject: [asterisk-users] How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P) Hello Asterisk users, We are having MANY but MANY problems configuring an analog fax machine to work properly on Asterisk, the first thing we do was to plug in the Fax analog machine to the FXS port of the Digium TDM410P, we set echocancel=no on zapata.conf and also faxdetect=yes on general section, but our Asterisk CRASH every time we try to send/receive fax! We are using Asterisk 1.4.21 and Zaptel 1.4.11; the card does not shows any interrupt in /proc/interrupts We also tried with a Sangoma B600 on this machine and the same result! Then we tried with the sangoma and digium card on another asterisk box, with Asterisk 1.4.30 and DAHDI 2.2.x and the fax was not reliable 100% but at least Asterisk vener went down We cant make any upgrade of Asterisk/Zaptel due to some rules of the customer, the do not want to use fax2email, they need to use the panasonic fax machine, this is driving me crazy! We also tried with a HT502 with passtrough fax mode and pcmu and pcma enabled and the same result, asterisk does down when trying to receive/send a fax What could solve our problem? what else should we try about configuration? just faxdetect=incoming and set echocancel=yes and that's all? Please your help, we really need to put this working Thanks in advance for all your help! -- Saludos Danny Dias SkypeID: danny.dias1 -- _ -- 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
