[asterisk-users] Jack in /usr/local/ means failure for asterisk
Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says XXX for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide PKG_CONFIG_PATH=/usr/local/lib64/pkgconfig so pkg-config --libs jack recognizes it. Modified /etc/ld.so.conf so libjack.so is cached in ld.so.cache: motie...@pbx3:/etc$ strings ld.so.cache | grep jack libjackserver.so.0 /usr/local/lib64/libjackserver.so.0 libjackserver.so /usr/local/lib64/libjackserver.so libjack.so.0 /usr/local/lib64/libjack.so.0 libjack.so /usr/local/lib64/libjack.so So... When I run in asterisk source dir: ./configure --disable-xmldoc, output has this line: checking for jack/jack.h... yes however, make menuselect shows XXX app_jack System information: motie...@pbx3:/usr/src/asterisk-1.6.2.7$ uname -a Linux pbx3 2.6.18.8-xenU #2 SMP Thu Apr 29 15:55:34 EEST 2010 x86_64 GNU/Linux motie...@pbx3:/usr/src/asterisk-1.6.2.7$ cat /etc/debian_version 5.0.4 Any help/suggestions how should I report Jack to asterisk really appreciated. Is it a bug? Should I report it that asterisk finds/does not find jack to mantis? Full configure log: http://paste.ubuntu.com/439830/ P.S. jack 0.109.2 (default debian repos) works fine. Motiejus -- _ -- 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] Jack in /usr/local/ means failure for asterisk
Motiejus Jakštys desired@gmail.com wrote: Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says XXX for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide PKG_CONFIG_PATH=/usr/local/lib64/pkgconfig so pkg-config --libs jack recognizes it. Modified /etc/ld.so.conf so libjack.so is cached in ld.so.cache: motie...@pbx3:/etc$ strings ld.so.cache | grep jack libjackserver.so.0 /usr/local/lib64/libjackserver.so.0 libjackserver.so /usr/local/lib64/libjackserver.so libjack.so.0 /usr/local/lib64/libjack.so.0 libjack.so /usr/local/lib64/libjack.so So... When I run in asterisk source dir: ./configure --disable-xmldoc, output has this line: checking for jack/jack.h... yes however, make menuselect shows XXX app_jack System information: motie...@pbx3:/usr/src/asterisk-1.6.2.7$ uname -a Linux pbx3 2.6.18.8-xenU #2 SMP Thu Apr 29 15:55:34 EEST 2010 x86_64 GNU/Linux motie...@pbx3:/usr/src/asterisk-1.6.2.7$ cat /etc/debian_version 5.0.4 Any help/suggestions how should I report Jack to asterisk really appreciated. Is it a bug? Should I report it that asterisk finds/does not find jack to mantis? Full configure log: http://paste.ubuntu.com/439830/ P.S. jack 0.109.2 (default debian repos) works fine. You need in your ./configure command line --prefix=/usr --mandir=/usr/share/man along with any other options and asterisk should see things. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Jack in /usr/local/ means failure for asterisk
Tried the following, both did not work: jack: ./configure --prefix=/usr make sudo make install ./configure --disable-xmldoc make menuselect - same problem (XXX app_jack) Jack installed in /usr/local/ ./configure --prefix=/usr/local make menuselect - same problem (XXX app_jack) ran make without changing (and having ability to change anything) in menuselect, but app_jack.so didn't appear: motie...@pbx3:/usr/src/asterisk-1.6.2.7$ ls -l `find | grep jack` -rw-r--r-- 1 motiejus motiejus 27378 2008-12-15 16:40 ./apps/app_jack.c -rw-r--r-- 1 motiejus motiejus 0 2010-05-26 15:12 ./apps/.app_jack.makeopts -rw-r--r-- 1 motiejus motiejus 166 2010-05-26 15:12 ./apps/.app_jack.moduleinfo motie...@pbx3:/usr/src/asterisk-1.6.2.7$ Any more suggestions? On Wed, May 26, 2010 at 2:21 PM, cov...@ccs.covici.com wrote: Motiejus Jakštys desired@gmail.com wrote: Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says XXX for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide PKG_CONFIG_PATH=/usr/local/lib64/pkgconfig so pkg-config --libs jack recognizes it. Modified /etc/ld.so.conf so libjack.so is cached in ld.so.cache: motie...@pbx3:/etc$ strings ld.so.cache | grep jack libjackserver.so.0 /usr/local/lib64/libjackserver.so.0 libjackserver.so /usr/local/lib64/libjackserver.so libjack.so.0 /usr/local/lib64/libjack.so.0 libjack.so /usr/local/lib64/libjack.so So... When I run in asterisk source dir: ./configure --disable-xmldoc, output has this line: checking for jack/jack.h... yes however, make menuselect shows XXX app_jack System information: motie...@pbx3:/usr/src/asterisk-1.6.2.7$ uname -a Linux pbx3 2.6.18.8-xenU #2 SMP Thu Apr 29 15:55:34 EEST 2010 x86_64 GNU/Linux motie...@pbx3:/usr/src/asterisk-1.6.2.7$ cat /etc/debian_version 5.0.4 Any help/suggestions how should I report Jack to asterisk really appreciated. Is it a bug? Should I report it that asterisk finds/does not find jack to mantis? Full configure log: http://paste.ubuntu.com/439830/ P.S. jack 0.109.2 (default debian repos) works fine. You need in your ./configure command line --prefix=/usr --mandir=/usr/share/man along with any other options and asterisk should see things. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Jack in /usr/local/ means failure for asterisk
Opened a bug report for it: https://issues.asterisk.org/view.php?id=17402 2010/5/26 Motiejus Jakštys desired@gmail.com: Tried the following, both did not work: jack: ./configure --prefix=/usr make sudo make install ./configure --disable-xmldoc make menuselect - same problem (XXX app_jack) Jack installed in /usr/local/ ./configure --prefix=/usr/local make menuselect - same problem (XXX app_jack) ran make without changing (and having ability to change anything) in menuselect, but app_jack.so didn't appear: motie...@pbx3:/usr/src/asterisk-1.6.2.7$ ls -l `find | grep jack` -rw-r--r-- 1 motiejus motiejus 27378 2008-12-15 16:40 ./apps/app_jack.c -rw-r--r-- 1 motiejus motiejus 0 2010-05-26 15:12 ./apps/.app_jack.makeopts -rw-r--r-- 1 motiejus motiejus 166 2010-05-26 15:12 ./apps/.app_jack.moduleinfo motie...@pbx3:/usr/src/asterisk-1.6.2.7$ Any more suggestions? On Wed, May 26, 2010 at 2:21 PM, cov...@ccs.covici.com wrote: Motiejus Jakštys desired@gmail.com wrote: Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says XXX for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide PKG_CONFIG_PATH=/usr/local/lib64/pkgconfig so pkg-config --libs jack recognizes it. Modified /etc/ld.so.conf so libjack.so is cached in ld.so.cache: motie...@pbx3:/etc$ strings ld.so.cache | grep jack libjackserver.so.0 /usr/local/lib64/libjackserver.so.0 libjackserver.so /usr/local/lib64/libjackserver.so libjack.so.0 /usr/local/lib64/libjack.so.0 libjack.so /usr/local/lib64/libjack.so So... When I run in asterisk source dir: ./configure --disable-xmldoc, output has this line: checking for jack/jack.h... yes however, make menuselect shows XXX app_jack System information: motie...@pbx3:/usr/src/asterisk-1.6.2.7$ uname -a Linux pbx3 2.6.18.8-xenU #2 SMP Thu Apr 29 15:55:34 EEST 2010 x86_64 GNU/Linux motie...@pbx3:/usr/src/asterisk-1.6.2.7$ cat /etc/debian_version 5.0.4 Any help/suggestions how should I report Jack to asterisk really appreciated. Is it a bug? Should I report it that asterisk finds/does not find jack to mantis? Full configure log: http://paste.ubuntu.com/439830/ P.S. jack 0.109.2 (default debian repos) works fine. You need in your ./configure command line --prefix=/usr --mandir=/usr/share/man along with any other options and asterisk should see things. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] Help with IP Routing
Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- 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] Help with IP Routing
Assume previous IP is LAN. Forwarding public IP ports to LAN is straighforward. However, with SIP headers you will (don't know H323) have to modify outgoing SIP headers: replace LAN ip with WAN ip. For callers you have to substitute RTP destination IP For callees you have to substitute RTP source IP. I`m afraid you will have to check more details here: http://www.ietf.org/rfc/rfc3261.txt Maybe client sends server it's own IP address? However, dumb header substitution + port range forwarding should work in all cases for SIP. On Wed, May 26, 2010 at 3:55 PM, Nivin Kumar nivinkuma...@yahoo.in wrote: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- 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] Help with IP Routing
Try a Cisco ASA. It will rewrite the headers if configured properly. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motiejus Jakštys Sent: 26 May 2010 14:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with IP Routing Assume previous IP is LAN. Forwarding public IP ports to LAN is straighforward. However, with SIP headers you will (don't know H323) have to modify outgoing SIP headers: replace LAN ip with WAN ip. For callers you have to substitute RTP destination IP For callees you have to substitute RTP source IP. I`m afraid you will have to check more details here: http://www.ietf.org/rfc/rfc3261.txt Maybe client sends server it's own IP address? However, dumb header substitution + port range forwarding should work in all cases for SIP. On Wed, May 26, 2010 at 3:55 PM, Nivin Kumar nivinkuma...@yahoo.in wrote: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- 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] Help with IP Routing
Is there a tool that will allow me to automatically change sip headers in realtime? --- On Wed, 26/5/10, Motiejus Jakštys desired@gmail.com wrote: From: Motiejus Jakštys desired@gmail.com Subject: Re: [asterisk-users] Help with IP Routing To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, 26 May, 2010, 1:17 PM Assume previous IP is LAN. Forwarding public IP ports to LAN is straighforward. However, with SIP headers you will (don't know H323) have to modify outgoing SIP headers: replace LAN ip with WAN ip. For callers you have to substitute RTP destination IP For callees you have to substitute RTP source IP. I`m afraid you will have to check more details here: http://www.ietf.org/rfc/rfc3261.txt Maybe client sends server it's own IP address? However, dumb header substitution + port range forwarding should work in all cases for SIP. On Wed, May 26, 2010 at 3:55 PM, Nivin Kumar nivinkuma...@yahoo.in wrote: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- 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] Help with IP Routing
Hello, if the remote side (the public IP side) is capable to do something like asterisk's nat=yes (in sip.conf), than a mascerading router (like every cheap DSL router) would do enough NAT do let SIP work. If the remote side does not support that nat-hack (which is not SIP standard), than you will need a NATing router also doing a lot of SIP header rewriting. Maybe the most easy thing will be to install asterisk on the NATing machine and operating regular SIP links on both sides. Roger. Nivin Kumar schrieb: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- 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] Help with IP Routing
- Nivin Kumar nivinkuma...@yahoo.in wrote: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? --- And this is related to Asterisk.. how? If your 'particular Windows based softswitch' doesn't in fact allow you to change the listening interfaces then it sounds like one great piece of software. If you're going to post something completely OT to the list, at least have the courtesy of telling us what softswitch you're talking about? --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] Help with IP Routing
Nivin Kumar schrieb: Is there a tool that will allow me to automatically change sip headers in realtime? Hi, imho changing the SIP headers will not be sufficient, since the old IP addresses are now private IP addresses (only in your network, outside, there are still public, but pointing not to your equipment). You will need a gateway, which does both: NAT 1:1, old IP addresses - new IP addresses and rewriting or all SIP headers, including those headers concerning the RTP endpoints. Maybe, you can do this with OpenSIPS. But I'm not sure about the SIP-headers for RTP. For H.323, it is imho less complicate, since it is robust for NAT and has no headers including IP addresses. Regards, Roger. -- _ -- 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] routing of calls
Hello everyone,
any help please
I have asterisk installed in our call centre with aheeva platform and
centos linux,
We have 2 access provider I have configured the
etc/asterisk/extensions.conf in order to do the routing of calls
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,Dial(Zap/*g2*/${EXTEN})
exten = s,n,Hangup();
G2 is for the second provider and g1 for the first provider even I
configured the extensios.conf I have some calls passed from g1 instead g2
Any help please will be appreciated
Thanks and Regards.
--
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Re: [asterisk-users] OT: Help with IP Routing
Didn't realize you were so sensitive. My apologies!
The switch in question is called VoipSwitch. It's ok...we use it mainly for
billing. Most of traffic is carried on Asterisk and handed off to this
voipswitch for billing purposes. I've added OT in the subject. I posted it
here because I know there is a big pool of highly skilled voip techies and I
thought I'd pick their brains.
-Nivin
--- On Wed, 26/5/10, Tim Nelson tnel...@rockbochs.com wrote:
From: Tim Nelson tnel...@rockbochs.com
Subject: Re: [asterisk-users] Help with IP Routing
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wednesday, 26 May, 2010, 1:46 PM
#yiv1740746078 p {margin:0;}
- Nivin Kumar nivinkuma...@yahoo.in wrote:
Hello,
I'm in a bit of a fix. We have a particular Windows based softswitch which is
has its SIP and H323 ports hardcoded to listen on a particular IP address. The
problem is that the ISP is having major issues and we can no longer depend on
them for service. The softswitch will not listen on any other IP address and
this can not be fixed. I was thinking of creating a NAT network wherein we will
forward all traffic from another public ip address to this server, however I'm
not sure how this will work. Do I need to modify the sip headers? Any thoughts
or suggestions?
---
And this is related to Asterisk.. how?
If your 'particular Windows based softswitch' doesn't in fact allow you to
change the listening interfaces then it sounds like one great piece of
software. If you're going to post something completely OT to the list, at least
have the courtesy of telling us what softswitch you're talking about?
--Tim
-Inline Attachment Follows-
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Re: [asterisk-users] Getting ghost transfer or music on hold
Yes, the both extensions are SIP. The problem to get the "core show channels" output its happen too fast, so I cant get the output at the moment of the call... I have the log of CLI output, with all types log enables (WARNING, NOTICE, DEBUG), but nothing of unusual in the log shows. Prince Singh escreveu: Are your extensions(who get the music between the calls) on SIP ? When the issue occurs, note the SIP peer account with which it is occurring Without hanging up, do a "core show channels" to see how many channels are present for that same SIP peer. If your are unable to identify this yourself, then mail the output of "core show channels" as a reply to this mail. The "core show channels" should be done WITHOUT hanging up the problematic extension -- Regards, Prince Singh Drishti-Soft Solutions Pvt Ltd W: http://www.drishti-soft.com B: http://blog.drishti-soft.com On Wed, May 26, 2010 at 8:35 AM, Fabiano Carlos Heringer b...@grupoheringer.com.br wrote: Hi Everybody, Im getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ... In some calls, i get an atxfer or musiconhold in the middle of call, or listening another call (like a cross line) without any intervention of the user. I got this error in about 3-10% of the calls, on a randomic times, and not pattern observed, just happens, and about 5-10 seconds the problem goes out. I cant identify nothing that can reproduce the error... Its happens using between SIP calls, or using external interface (Digital Trunk). Got Ideas? 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 -- _ -- 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] OT: Windows TAPI command-line driver
Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app with the phone number as argument. ex when clicking on 555-555-: the TAPI driver would call customapp.exe 555-555- Regards, Michael -- _ -- 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] routing of calls
salaheddine elharit wrote: G2 is for the second provider and g1 for the first provider even I configured the extensios.conf I have some calls passed from g1 instead g2 Any help please will be appreciated Maybe if you asked a question, something could help. But, as it is stated now, I'm have no idea as to what you want help with. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] routing of calls
Doug, did you cancel your psychic friend's subscription? All programmers are supposed to be able to determine intent without full information :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, May 26, 2010 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] routing of calls salaheddine elharit wrote: G2 is for the second provider and g1 for the first provider even I configured the extensios.conf I have some calls passed from g1 instead g2 Any help please will be appreciated Maybe if you asked a question, something could help. But, as it is stated now, I'm have no idea as to what you want help with. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] routing of calls
Danny Nicholas wrote: Doug, did you cancel your psychic friend's subscription? All programmers are supposed to be able to determine intent without full information :) I had too! I'm on a budget and it was costing me more then my cable bill. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] OT: Windows TAPI command-line driver
GIYF - try this link http://www.voip-info.org/wiki/view/Asterisk+TAPI _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, May 26, 2010 10:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OT: Windows TAPI command-line driver Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app with the phone number as argument. ex when clicking on 555-555-: the TAPI driver would call customapp.exe 555-555- Regards, Michael -- _ -- 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] OT: Windows TAPI command-line driver
On 26 May 2010 15:59, Mike l...@virtutel.ca wrote: Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. There is a command-line tool dialer.exe that comes with Windows that allows you to dial through TAPI: dialer.exe 555-555- as you suggest. There would be no reason to use this in Outlook though as Outlook can call into the TAPI subsystem natively, and does so by default if you correctly configure your Phones and Modems under Control panel, and have a suitable TAPI module configured to talk to your PABX. If you are missing the TAPI component, Google Xtelsio or Activa TSP 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] OT: Windows TAPI command-line driver
Thanks, will take a look. Althought none of those things seem to allow me to call up my own handler for calls, does it? Or am I misreading? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, May 26, 2010 11:06 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] OT: Windows TAPI command-line driver GIYF - try this link http://www.voip-info.org/wiki/view/Asterisk+TAPI _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, May 26, 2010 10:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OT: Windows TAPI command-line driver Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app with the phone number as argument. ex when clicking on 555-555-: the TAPI driver would call customapp.exe 555-555- Regards, Michael -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
Hello I'm trying to install Dahdi through source code on a Fedora 13 host to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv isn't happy. 1. After successfully running make all; make install; make config, I edited /etc/dahdi/system.conf thusly: loadzone=fr defaultzone=fr fxsks=1 2. Then ran dahdi_cfg -vv which says: - DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) - 3. So I ran lscpi -v: 03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 20 I/O ports at a000 [size=256] Memory at e200 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel driver in use: netjet Kernel modules: wctdm, hisax, netjet FWIW, when I run modprobe wctdm followed by lsmod: # lsmod Module Size Used by wctdm 31892 0 dahdi 180789 1 wctdm netjet 12563 0 isdnhdlc3343 1 netjet crc_ccitt 1217 2 dahdi,isdnhdlc mISDNipac 28346 1 netjet mISDN_core 61414 3 netjet,mISDNipac I'm not sure whether I should use the wctdm driver or this netjet driver which I've never seen before. Could it be that dahdi_genconf modules added some ISDN-related items that I don't need? Thank you. -- _ -- 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] Better AMD module
Has anyone written a better AMD than the default AMD? The existing AMD works great but it has a few shortcomings... I do know about Sangoma but am just looking for a better AMD module. Thanks, John -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Wed, 26 May 2010 17:17:08 +0200, Vincent codecompl...@free.fr wrote: I'm trying to install Dahdi through source code on a Fedora 13 host to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv isn't happy. More information, as I investigate: # vi /etc/modprobe.d/dahdi.blacklist.conf #blacklist wct4xxp #blacklist wcte12xp #blacklist wct1xxp #blacklist wcte11xp #blacklist wctdm24xxp #blacklist wcfxo blacklist wctdm #blacklist wctc4xxp #blacklist wcb4xxp # /etc/init.d/dahdi stop Unloading DAHDI hardware modules: done # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: [ OK ] /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] -- _ -- 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] Help with IP Routing
Skip the whole NAT scenario. Put up an asterisk box with two network interfaces. One interface connects to the real world on your new IP address from your new ISP. The other interface can be on the same subnet as the windows box that you can't change. Set up a SIP trunk to your Windows box. Use packet 2 packet bridging in asterisk. Now that the emergency is over you can migrate off of your Windows thing at a more comfortable pace. You will be using someone else's public IP privately for awhile, but the main thing affected by that is your asterisk box won't be able to talk to anybody in that subnet in the outside world. You'll have to determine how bad of a thing that would be. BTW: What the heck is this software? Sounds like whoever wrote that wasn't thinking ahead. Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- 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] routing of calls
Hello All
i have set all extensions for 2 providers in dialplan.conf and
extensions.conf
the problem is all numbers take the same provider
when i change the g1 with g2 all the phones numbers take the secend
provider
; Outbound dial context
[aheeva_ccs]
; If we are dialing out through another Asterisk, sometimes when a call is
not
; answered the DIALSTATUS gets set to CANCEL and Asterisk just aborts the
DIAL
; and jumps directly to the h extension without continuing processing in the
; dialplan after the Dial application, which means that we do not send the
; DIALSTATUS to the CCS server after the dial. This is why we need to
capture
; here in the h extension and send a NOANSWER.
exten = h,1,NoOp(ds= ${DIALSTATUS});
exten = h,2,GotoIf($[${DIALSTATUS} = ANSWER]?6:3)
exten = h,3,GotoIf($[${DIALSTATUS} = CANCEL]?4:5)
exten =
h,4,AHEventsProxy(NOANSWER:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten =
h,5,AHEventsProxy(MSG_TYPE_TERMINATE_CALL:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH}:${AH_AGENTID})
exten = h,6,Hangup
exten = _OUT.,1,NoOp(AHEEVA1 Variables:
AH_PHONE_NUMBER=[${AH_PHONE_NUMBER}] AH_QUEUE=[${AH_QUEUE}]
AH_URL=[${AH_URL}] AH_RECORDID=[${AH_RECORDID}]
AH_AMD_REQUIRED=[${AH_AMD_REQUIRED}] AH_CALLERID=[${AH_CALLERID}]
AHEEVA_TRACKNUM=[${AHEEVA_TRACKNUM}] AH_LEAVE_MESSAGE=[${AH_LEAVE_MESSAGE}])
exten = _OUT.,2,SetCallerId(${AH_CALLERID})
exten = _OUT.,3,Dial(Zap/g1/${AH_PHONE_NUMBER},30)
exten = _OUT.,4,NoOp(Dial Status=[${DIALSTATUS}] Hangup
Cause=[${HANGUPCAUSE}])
exten = _OUT.,5,GotoIf($[${DIALSTATUS} = CHANUNAVAIL ${HANGUPCAUSE}
= 16]?6:8)
exten =
_OUT.,6,AHEventsProxy(MSG_TYPE_CALL_SIT:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten = _OUT.,7,Goto(9)
exten =
_OUT.,8,AHEventsProxy(${DIALSTATUS}:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten = _OUT.,9,NoOp()
thanks a lot
2010/5/26 Doug Lytle supp...@drdos.info
salaheddine elharit wrote:
G2 is for the second provider and g1 for the first provider even I
configured the extensios.conf I have some calls passed from g1
instead g2
Any help please will be appreciated
Maybe if you asked a question, something could help. But, as it is
stated now, I'm have no idea as to what you want help with.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Libpri 1.4.11 Released
The Asterisk Development Team has announced the release of version 1.4.11 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ This release contains many fixes and new features, among them being: 1.) Support for NT-PTMP BRI links, including support for multiple TEIs and connecting of BRI phones. 2.) Support for allowing persistent Q.921 drops on both NT and TE PTMP links, as well as automatically requesting that Q.921 data links reactivate when needed by Q.931. 3.) T309 is enabled by default. 4.) Problems with Keypad Facility Digits were addressed. 5.) A number of additional service related features were added: Connected Line Information, HOLD/RELEASE support, Call Deflection/Call Rerouting, as well as partial subaddress support. They are supported in the Q.SIG and EuroISDN switch types, and most currently require using the trunk version of Asterisk. 6.) Many potential and realized Q.921 related problems, particularly during retransmissions and other scenarios involving medium to high packet loss. For a full list of changes in the current release candidates, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11 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] Jack in /usr/local/ means failure for asterisk
Hi Motiejus! If all else fails for the moment, it should be quite simple to move JACK. Move all jack applications from /usr/local/bin to /usr/bin. In /usr/local/lib move the dir jack and libjack* to /usr/lib. That should be it for the moment. another thing is to hack the JACK confiugre script. I did that as well. As JACK didn't like to install, while a system wide installation was there. I hope this can help a bit. Question: Do you also use app_jack in a dialplan from the CLI? How does that work (reliability wise)? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] routing of calls
I dont know, maybe I am missing it. I see nothing off the top of my head that
shows you attempting to dial out 2 different providers or fail between them.
Both times you have posted code I see a dial command set to go to a single Zap
Group, and no failure code or Prefix that determines how or when to dial the
other Zap Group instead. I think your getting lost in your code, or are
missing things you should be providing to the mail list so we can figure out
the problem for you.
WHAT is your determining factor for dialing Group1 or Group2? Does Group 1
dial with a 8 prefix and Group 2 dial with a 9 prefix? Are you attempting to
failover from Group 1 to Group 2 when you get a cancel dialstatus. Also your
dialstatus getting set to cancel should be your user deciding to hangup the
call. I dial between asterisk servers all the time, and have used some as
proxy's to resolve weird provider issues, I haven't seen a cancel just randomly
showup in place of a valid DIALSTATUS when doing so, without the agent/user
canceling the call. However I obviously have not tested this against every
version like between 1.2-1.6 (I have however done 1.2-1.4 and 1.4-1.6.).
--
Trevor Benson
dCAP, LPIC-1, CLA, Network+, MCP, CNA
A1 Networks - Network Engineer
DID (707)703-1041
FAX (707)703-1983
On May 26, 2010, at 8:41 AM, salaheddine elharit wrote:
Hello All
i have set all extensions for 2 providers in dialplan.conf and extensions.conf
the problem is all numbers take the same provider
when i change the g1 with g2 all the phones numbers take the secend provider
; Outbound dial context
[aheeva_ccs]
; If we are dialing out through another Asterisk, sometimes when a call is not
; answered the DIALSTATUS gets set to CANCEL and Asterisk just aborts the DIAL
; and jumps directly to the h extension without continuing processing in the
; dialplan after the Dial application, which means that we do not send the
; DIALSTATUS to the CCS server after the dial. This is why we need to capture
; here in the h extension and send a NOANSWER.
exten = h,1,NoOp(ds= ${DIALSTATUS});
exten = h,2,GotoIf($[${DIALSTATUS} = ANSWER]?6:3)
exten = h,3,GotoIf($[${DIALSTATUS} = CANCEL]?4:5)
exten =
h,4,AHEventsProxy(NOANSWER:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten =
h,5,AHEventsProxy(MSG_TYPE_TERMINATE_CALL:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH}:${AH_AGENTID})
exten = h,6,Hangup
exten = _OUT.,1,NoOp(AHEEVA1 Variables: AH_PHONE_NUMBER=[${AH_PHONE_NUMBER}]
AH_QUEUE=[${AH_QUEUE}] AH_URL=[${AH_URL}] AH_RECORDID=[${AH_RECORDID}]
AH_AMD_REQUIRED=[${AH_AMD_REQUIRED}] AH_CALLERID=[${AH_CALLERID}]
AHEEVA_TRACKNUM=[${AHEEVA_TRACKNUM}] AH_LEAVE_MESSAGE=[${AH_LEAVE_MESSAGE}])
exten = _OUT.,2,SetCallerId(${AH_CALLERID})
exten = _OUT.,3,Dial(Zap/g1/${AH_PHONE_NUMBER},30)
exten = _OUT.,4,NoOp(Dial Status=[${DIALSTATUS}] Hangup
Cause=[${HANGUPCAUSE}])
exten = _OUT.,5,GotoIf($[${DIALSTATUS} = CHANUNAVAIL ${HANGUPCAUSE}
= 16]?6:8)
exten =
_OUT.,6,AHEventsProxy(MSG_TYPE_CALL_SIT:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten = _OUT.,7,Goto(9)
exten =
_OUT.,8,AHEventsProxy(${DIALSTATUS}:${AHEEVA_TRACKNUM}:${AH_PHONE_NUMBER}:${AH_RECORDID}:${EPOCH})
exten = _OUT.,9,NoOp()
thanks a lot
2010/5/26 Doug Lytle supp...@drdos.info
salaheddine elharit wrote:
G2 is for the second provider and g1 for the first provider even I
configured the extensios.conf I have some calls passed from g1
instead g2
Any help please will be appreciated
Maybe if you asked a question, something could help. But, as it is
stated now, I'm have no idea as to what you want help with.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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[asterisk-users] ring splash
Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, 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] Better AMD module
What version of asterisk are you running. What shortcomings are you experiencing in AMD? What type of tuning have you done or settings are you using with AMD? What are you doing after you run AMD on the call? If the call is human you do X if its not you do Y? Are these AGI's or Goto's or??? -- Trevor Benson dCAP, LPIC-1, CLA, Network+, MCP, CNA A1 Networks - Network Engineer DID (707)703-1041 FAX (707)703-1983 On May 26, 2010, at 8:24 AM, John Rose wrote: Has anyone written a better AMD than the default AMD? The existing AMD works great but it has a few shortcomings… I do know about Sangoma but am just looking for a better AMD module. Thanks, John -- _ -- 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] ring splash
On 05/26/2010 11:36 AM, Jeff LaCoursiere wrote: So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) The simple answer is no; the ling rings, the ringing is detected and forwarded up the software stack. The more complex answer is that first, the TDM410P doesn't actually know anything about detecting ringing, ring patterns, or anything of the like, it's essentially dumb hardware :-) The driver for the hardware detects the incoming ring voltage and debounces it before reporting it to the DAHDI core and then upstream to Asterisk; it is possible you could set the debounce timer to require that the ring last at least 500ms (or maybe even a full second) before reporting it, which would absorb these ring splashes. -- 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] ring splash
The ring splash is a long standing feature of call forwarding. Of course somewhere in the Asterisk code a change could be made to extend the time required to detect a valid ring. But, how about just unplugging the pots lines from the PBX with a quick restore ability? Unplug lines at the NID, or open bridging clips or whatever applies. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, May 26, 2010 11:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ring splash Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, 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 -- _ -- 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] VoIP over virtualized VPN
Hi List, Our company has several small distributed offices we would like to inter-connect with bridged VPN a single subnet (last example in http://www.shorewall.net/OPENVPN.html). We have SIP phones in every office (up to 5) so we can use SIP without any NATing and securely. Max theoretical simultaneous calls possible ~30, but we have ~5-10 @ regular basis. OpenVPN server would be in the same datacenter like Asterisk PBX (in one physical subnet). Asterisk and OpenVPN are virtualized XEN guests. I wonder about overheads, system loads and other possible gotchas in this setup. Is there anything I should (re-)consider before implementing this? Anyone had difficulties running VoIP or VPN traffic over (virtualized if it makes any difference) VPN? We use mainly g729 and speex, and very little g711. Regards Motiejus -- _ -- 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] ring splash
Just set the POTS lines to answer after a second ring rather than after the first. Problem solved. On 5/26/2010 11:36 AM, Jeff LaCoursiere wrote: Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, j -- Brent Davidson Texas Country Title Company 112 W 2nd / P.O. Box 663 Cameron, TX 76520 254-605-0140 ex. 21 br...@texascountrytitle.com -- _ -- 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] ring splash
Strangely enough I have used this many times with our POTS from ATT. We get ring splash, but didnt get a ghost ring into the system, just the valid ring that was redirected to the VoIP lines after forwarding. Although I think i had the Asterisk-GUI creating the dialplan on these systems, not sure if it changes anything besides the dialplan though that would cause it to seem normal during a forward of a POTS line. -- Trevor Benson dCAP, LPIC-1, CLA, Network+, MCP, CNA A1 Networks - Network Engineer DID (707)703-1041 FAX (707)703-1983 On May 26, 2010, at 9:43 AM, Cary Fitch wrote: The ring splash is a long standing feature of call forwarding. Of course somewhere in the Asterisk code a change could be made to extend the time required to detect a valid ring. But, how about just unplugging the pots lines from the PBX with a quick restore ability? Unplug lines at the NID, or open bridging clips or whatever applies. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, May 26, 2010 11:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ring splash Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, 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 -- _ -- 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] VoIP over virtualized VPN
I use openvpn for VOIP traffic all the time. It's not a commercial application, and only one simultaneous call usually on each vpn link, but I even have a VPN client on a Linksys WRT-54g wireless router with 1 phone behind it - it works flawlessly, so it does not take a lot of CPU to run a vpn connection. Andrew 2010/5/26 Motiejus Jakštys desired@gmail.com: Hi List, Our company has several small distributed offices we would like to inter-connect with bridged VPN a single subnet (last example in http://www.shorewall.net/OPENVPN.html). We have SIP phones in every office (up to 5) so we can use SIP without any NATing and securely. Max theoretical simultaneous calls possible ~30, but we have ~5-10 @ regular basis. OpenVPN server would be in the same datacenter like Asterisk PBX (in one physical subnet). Asterisk and OpenVPN are virtualized XEN guests. I wonder about overheads, system loads and other possible gotchas in this setup. Is there anything I should (re-)consider before implementing this? Anyone had difficulties running VoIP or VPN traffic over (virtualized if it makes any difference) VPN? We use mainly g729 and speex, and very little g711. Regards Motiejus -- _ -- 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] Libpri 1.4.11 Released
Thanks for the update. How to upgrade to the latest stable release without compliling Asterisk again? Can you please explain and detail the commands? We are running PBXinaFlash with LibPRI 1.4.10.1 which gives lots of problems. Thanks On Wed, May 26, 2010 at 12:27 PM, Asterisk Development Team asteriskt...@digium.com wrote: The Asterisk Development Team has announced the release of version 1.4.11 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ This release contains many fixes and new features, among them being: 1.) Support for NT-PTMP BRI links, including support for multiple TEIs and connecting of BRI phones. 2.) Support for allowing persistent Q.921 drops on both NT and TE PTMP links, as well as automatically requesting that Q.921 data links reactivate when needed by Q.931. 3.) T309 is enabled by default. 4.) Problems with Keypad Facility Digits were addressed. 5.) A number of additional service related features were added: Connected Line Information, HOLD/RELEASE support, Call Deflection/Call Rerouting, as well as partial subaddress support. They are supported in the Q.SIG and EuroISDN switch types, and most currently require using the trunk version of Asterisk. 6.) Many potential and realized Q.921 related problems, particularly during retransmissions and other scenarios involving medium to high packet loss. For a full list of changes in the current release candidates, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11 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 -- _ -- 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] Getting 'username' of sip peer
Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from 'TestSIPUser' then I want to be able to get the value 'testuser' Is it possible to get the value of 'username' of the peer in the dialplan using some application/function ? Thanks, Deepesh -- _ -- 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] Getting 'username' of sip peer
I might be wrong, but I think that adding fullname=xxx to the context will populate CALLERID(name) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D Sent: Wednesday, May 26, 2010 12:18 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting 'username' of sip peer Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from 'TestSIPUser' then I want to be able to get the value 'testuser' Is it possible to get the value of 'username' of the peer in the dialplan using some application/function ? Thanks, Deepesh -- _ -- 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] VoIP over virtualized VPN
On Wed, May 26, 2010 at 8:01 PM, Andrew Hakman andrew.hak...@gmail.com wrote: I use openvpn for VOIP traffic all the time. It's not a commercial application, and only one simultaneous call usually on each vpn link, but I even have a VPN client on a Linksys WRT-54g wireless router with 1 phone behind it - it works flawlessly, so it does not take a lot of CPU to run a vpn connection. Andrew Thanks for the answer, but I am am asking about larger setups (30 phones). -- _ -- 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] VoIP over virtualized VPN
I have several Atom based boxes running OpenVPN and processing up to six simultaneous calls over it with no issues. I am quite sure it could do more. Load is still at .2 :) j On Wed, 26 May 2010, Andrew Hakman wrote: I use openvpn for VOIP traffic all the time. It's not a commercial application, and only one simultaneous call usually on each vpn link, but I even have a VPN client on a Linksys WRT-54g wireless router with 1 phone behind it - it works flawlessly, so it does not take a lot of CPU to run a vpn connection. Andrew 2010/5/26 Motiejus Jakštys desired@gmail.com: Hi List, Our company has several small distributed offices we would like to inter-connect with bridged VPN a single subnet (last example in http://www.shorewall.net/OPENVPN.html). We have SIP phones in every office (up to 5) so we can use SIP without any NATing and securely. Max theoretical simultaneous calls possible ~30, but we have ~5-10 @ regular basis. OpenVPN server would be in the same datacenter like Asterisk PBX (in one physical subnet). Asterisk and OpenVPN are virtualized XEN guests. I wonder about overheads, system loads and other possible gotchas in this setup. Is there anything I should (re-)consider before implementing this? Anyone had difficulties running VoIP or VPN traffic over (virtualized if it makes any difference) VPN? We use mainly g729 and speex, and very little g711. Regards Motiejus -- _ -- 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] Getting 'username' of sip peer
On Wed, 2010-05-26 at 22:48 +0530, Deepesh D wrote:
When a call is made from any of these peers I want to get the username
of the peer.
for eg:- If a call is being made from 'TestSIPUser' then I want to be
able to get the value 'testuser'
I can think of two ways of doing this. The first is to use the
SIPCHANINFO() dialplan function, like this:
exten=123,1,Verbose(0,The call came from ${SIPCHANINFO(peername)})
The other option is to use the setvar=variable=value setting in the
peer definition in sip.conf. For example, if you add
setvar=USERID=jsmith in a user/peer/friend definition, Asterisk would
automagically create a channel variable named USERID with a value of
jsmith every time this device made a call into Asterisk.
--
Jared Smith
Sr. Trainer
Digium, Inc.
--
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Re: [asterisk-users] Better AMD module
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Trevor Benson What version of asterisk are you running. 1.6.0.28. What shortcomings are you experiencing in AMD? Was thinking that an asynchronous AMD that runs longer and sends AMI UserEvents would be better. The existing AMD is synchronous and does a fast detection within a few seconds. Also was thinking that generic answering machine beep detection would help, using an FFT or existing Asterisk dsp.c functions. And AMD on a non-T.38 call won't report fax, modem or SIT answerers. (I realize SIT audio comes in before answer) Also for some reason I need to play some audio out to the carrier to get RTP coming in otherwise AMD always reports an erroneous result. What type of tuning have you done or settings are you using with AMD? I've messed around some. I haven't run it live yet across multiple carriers and am worried about accuracy. What are you doing after you run AMD on the call? If the call is human you do X if its not you do Y? Are these AGI's or Goto's or??? FastAGI from C#. If AMD produces a MACHINE result I play a file that pauses if it is interrupted by received audio and restarts doing this x number of times until the message is laid onto the voicemail recorder. If HUMAN I assume human and prompt for DTMF etc.. So I am looking for a better AMD.c if there is one available... Thanks, John Rose j...@westfax.com -- Trevor Benson dCAP, LPIC-1, CLA, Network+, MCP, CNA A1 Networks - Network Engineer DID (707)703-1041 FAX (707)703-1983 On May 26, 2010, at 8:24 AM, John Rose wrote: Has anyone written a better AMD than the default AMD? The existing AMD works great but it has a few shortcomings... I do know about Sangoma but am just looking for a better AMD module. Thanks, John -- -- _ -- 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] ring splash
On Wed, 26 May 2010, Brent Davidson wrote: Just set the POTS lines to answer after a second ring rather than after the first. Problem solved. Now that sounds like a good plan. But a quick look through the options in zapata.conf don't show any kind of option for waiting before pickup. Something that *did* look promising is distinctive ring detection. Has anyone used this ability to detect different ring styles? Presumably with a lot of trial and error I might be able to detect a ring splash from a real ring. ALternatively if someone knows how to actually make the card wait X rings or seconds before answering, that would be great. I'm coming up zero on searches. Its already set to wait for callerid, so I am a bit confused why it is picking up on a splash... seems it should wait for that second ring anyway. Cheers, j On 5/26/2010 11:36 AM, Jeff LaCoursiere wrote: Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, j -- Brent Davidson Texas Country Title Company 112 W 2nd / P.O. Box 663 Cameron, TX 76520 254-605-0140 ex. 21 br...@texascountrytitle.com -- _ -- 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] ring splash
- Jeff LaCoursiere j...@jeff.net wrote: On Wed, 26 May 2010, Brent Davidson wrote: Just set the POTS lines to answer after a second ring rather than after the first. Problem solved. Now that sounds like a good plan. But a quick look through the options in zapata.conf don't show any kind of option for waiting before pickup. It would be in your dialplan. (Untested, OTMH, etc) Dialplan: [from-analog-lines] exten = s,1,Wait(2) exten = s,n,Answer() exten = s,n,Play(tt-monkeys) exten = h,1,Hangup() Again, that is untested, just off the top of my head. The key is putting a wait before your Answer(). A phantom ring/ring splash should fade away before the Wait() period is finished, therefore not hitting your Answer() or Dial() or whatever you have causing all sorts of panic and grief. :-) --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] Music on Hold
Pressing hold on the telephone set may not be sending hold to Asterisk to trigger the correct action. You can verify this from the CLI. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign https://signup.live.com/signup.aspx?id=60969 up now. -- _ -- 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] Music on Hold
Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] call droped if second caller enter meetme conference
Hello Group, some strange problem i have on my setup. If a second caller entering a meetme conference dropping the first one. my setup is using asterisk 1.6.2.6 and dahdi 2.1.1.1 with realtime, the conference room numbers storing in a mysql database. the calls came from a sip provider. there are nothing in logfiles with sip debug on :( Has anyone the same problem and a solution for me? Thanx for all. Daniel Knoll -- _ -- 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] Extension state can get stuck in 'Ringing' state
Hi, I've noticed that if a phone goes UNREACHABLE while it is Ringing, when the phone comes back, Asterisk will not clear the channel that was created, so it still thinks it is in the Ringing state. The only way to clear this is to do a soft hangup on the SIP channel or to restart Asterisk. Unfortunately these issues are very hard to automatically track down and clear and it seems like if a phone goes UNREACHABLE, Asterisk should clear the channel anyways. This is at 1.4.26.2. I'm planning to upgrade to 1.4.31 shortly. I will see if I can replicate the problem in that version as well. Has anyone else noticed this? Thanks. -- James -- _ -- 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] Music on Hold
Hello Ya you are right moh in Asterisk is not triggered. Is there any solution to that ? Regards Taimur Hasan -THQ- !!!ONE From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 26 May 2010 13:39:00 -0500 Subject: Re: [asterisk-users] Music on Hold Pressing “hold” on the telephone set may not be sending “hold” to Asterisk to trigger the correct action. You can verify this from the CLI. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] Jack in /usr/local/ means failure for asterisk
On Wed, May 26, 2010 at 7:28 PM, Julien Claassen jul...@c-lab.de wrote: Hi Motiejus! If all else fails for the moment, it should be quite simple to move JACK. Move all jack applications from /usr/local/bin to /usr/bin. In /usr/local/lib move the dir jack and libjack* to /usr/lib. That should be it for the moment. another thing is to hack the JACK confiugre script. I did that as well. As JACK didn't like to install, while a system wide installation was there. I hope this can help a bit. Thanks Julien! I am eager to try that, but with a question. Why should I hack the installation script if jack is already installed and libraries moved? Please explain that in more detail :-) If you mean pre-hacking (before installation) (and no library/binary movement after hacking if I understand correctly), could you please share the script (possibly a diff against original?) Question: Do you also use app_jack in a dialplan from the CLI? How does that work (reliability wise)? This is my wiki entry. This works, but has three major disadvantages: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial#DialtogetherwithopeniningJackportsforcal 1. There is a weird sound for ~0.3 second observed when jack ports connect (like BLUAH hello). It isn't major (after asking several customers barely noticed anything). However, I can hear this clearly when I am calling; 2. Early RTP (pre-answer) cannot be transferred to jack in this manner (for example, when calling to mobiles or landlines you cannot give jack operator messages). 3. Not much tested in production environment - pretty unstable. See below. Because of the (2) reason I am planning to execute this line: *CLI core set chanvar SIP/$channel JACK_HOOK(manipulate,i(SIP/$channel:input),o(SIP/$channel:output)) on Through AMI. Executing it through CLI is tested and it worked without problems (including pre-answering). I will write a C++ script that does the AMI observing and jack connecting through AMI, I will push thos to SoundPatty. About reliability... I don't know what (is it my program or old jackd), but something causes deadlocks for jackd and both asterisk and jackd hang until I press ctrl-c for my jackd server (then asterisk gets back to life). This is the reason I am upgrading. One notice for deploying jack_hook+asterisk: be sure to disconnect and remove jack client before quitting the program (thread). This fixed some deadlocking my past observings. Regards Motiejus -- _ -- 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] Music on Hold
Either teach your operators to press a new sequence that will send hold to asterisk or reprogram your phone. I know how to do with Polycom phones, but not linksys. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:46 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold Hello Ya you are right moh in Asterisk is not triggered. Is there any solution to that ? Regards Taimur Hasan -THQ- !!!ONE _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 26 May 2010 13:39:00 -0500 Subject: Re: [asterisk-users] Music on Hold Pressing hold on the telephone set may not be sending hold to Asterisk to trigger the correct action. You can verify this from the CLI. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. https://signup.live.com/signup.aspx?id=60969 _ Hotmail: Free, trusted and rich email service. Get it now. https://signup.live.com/signup.aspx?id=60969 -- _ -- 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] Music on Hold
You would need to see if there is a hook flash hold. Try playing with a hook/flash ( ie do a flash, wait, then hangup phone, it may send the onhold message ) it may also ring back. Or you will have to park the call Hook flash , Dial 700 (if that's your park extension), hangup, then recall the call if you want to answer it. An analog port is usefully on hook or off hook, no on hold unless the ATA has something documented for it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 2:46 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold Hello Ya you are right moh in Asterisk is not triggered. Is there any solution to that ? Regards Taimur Hasan -THQ- !!!ONE _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 26 May 2010 13:39:00 -0500 Subject: Re: [asterisk-users] Music on Hold Pressing hold on the telephone set may not be sending hold to Asterisk to trigger the correct action. You can verify this from the CLI. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan Sent: Wednesday, May 26, 2010 1:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. https://signup.live.com/signup.aspx?id=60969 _ Hotmail: Free, trusted and rich email service. Get it now. https://signup.live.com/signup.aspx?id=60969 -- _ -- 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] [Dahdi] DAHDI_CHANCONFIG failed on channel 1?
On Wed, 26 May 2010 17:30:08 +0200, Vincent codecompl...@free.fr wrote: More information, as I investigate: For those having the same issue, here's what I learned: 1. In /etc/modprobe.d/dahdi.blacklist.conf, blacklist the netjet driver: blacklist netjet 2. To configure Dahdi, edit /etc/dahdi/system.conf: #For France loadzone= fr defaultzone = fr fxsks = 1 Next, start Dahdi... /etc/init.d/dahdi start ... and check /var/log/messages. DON'T RUN dahdi_genconf, as it overwrites system.conf. == I still have a couple of issues left: 1. When I run dahdi_genconf: /usr/sbin/dahdi_genconf: Failed to open /etc/asterisk/dahdi-channels.conf: No such file or directory 2. /etc/init.d/dahdi start: Loading DAHDI hardware modules: wctdm: [ OK ] /usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting Running dahdi_cfg: [ OK ] Thank you. -- _ -- 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] Jack in /usr/local/ means failure for asterisk
Hi Motiejus! I of course menat prehacking the installation script. I can unpack my jack-tarball or get a new svn and see, what I did. It was some time ago. And I usually don't have to hack the installation prefix. Information: I'm running my asterisk and JACK on a simple desktop system (no GUI though! :-) ). It's more like a toy or convenience. I can implement an answering machine with it, that I can use very well - I'm blind. Also I'd like to emply asterisk to make googletalk calls. I just use the asterisk CLI for my calls or a simple bash script, that eases the usage a bit, when calling (these commands can be rather long. :-) but I haven't used asterisk for quite a while (tested it half a year ago and wasn't happy). But I like to rebuild it, because it was so useful and nice to talk with people all over the planet. :-) I suppose you wouldn't be interested in a simple c program, that can be used from within a dialplan to let it ring and another hack to pick up the phone, without having a softphone? Kindly yours Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] ring splash
On 5/26/2010 1:16 PM, Tim Nelson wrote: - Jeff LaCoursierej...@jeff.net wrote: On Wed, 26 May 2010, Brent Davidson wrote: Just set the POTS lines to answer after a second ring rather than after the first. Problem solved. Now that sounds like a good plan. But a quick look through the options in zapata.conf don't show any kind of option for waiting before pickup. It would be in your dialplan. (Untested, OTMH, etc) Dialplan: [from-analog-lines] exten = s,1,Wait(2) exten = s,n,Answer() exten = s,n,Play(tt-monkeys) exten = h,1,Hangup() Again, that is untested, just off the top of my head. The key is putting a wait before your Answer(). A phantom ring/ring splash should fade away before the Wait() period is finished, therefore not hitting your Answer() or Dial() or whatever you have causing all sorts of panic and grief. :-) --Tim I was thinking there was a way to directly set the number of rings before the system picked up the call, but it looks like Tim is right. The Wait statement before the answer appears to be the only way to handle this. I actually used this technique to deal with some phantom rings that were occurring at one of my branch offices. The Telco had the switch set up to periodically test the line (like every 30 minutes) and Asterisk was detecting those test pulses as a ring and answering the call, then passing it on into the operator queue before the system could detect the hang-up. The poor lady at that office nearly had a nervous breakdown before I figured out how to filter out the phantom calls with the wait command. -- _ -- 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] Music on Hold
hello, which phone do you have behind the pap2 cause the hook flash time sometimes could be set in the phone and then it will work with the pap2 also. you should have a look at spaconfig.de (its a german website) but the default parameters in sip and regional conf, may help you. best regards steve taimur hasan schrieb: Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan *-THQ- !!!ONE* Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. https://signup.live.com/signup.aspx?id=60969 -- _ -- 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] Jack in /usr/local/ means failure for asterisk
Hi Julien, yes, I am thinking about implementing something better than asterisk, but it is a future talk. It really lacks support for flexibility :S I would be grateful if you found the hacked configure script and sent it to me :-) I did not really understand the part about C program, could you explain in more detail? Best Motiejus On Wed, May 26, 2010 at 11:19 PM, Julien Claassen jul...@c-lab.de wrote: Hi Motiejus! I of course menat prehacking the installation script. I can unpack my jack-tarball or get a new svn and see, what I did. It was some time ago. And I usually don't have to hack the installation prefix. Information: I'm running my asterisk and JACK on a simple desktop system (no GUI though! :-) ). It's more like a toy or convenience. I can implement an answering machine with it, that I can use very well - I'm blind. Also I'd like to emply asterisk to make googletalk calls. I just use the asterisk CLI for my calls or a simple bash script, that eases the usage a bit, when calling (these commands can be rather long. :-) but I haven't used asterisk for quite a while (tested it half a year ago and wasn't happy). But I like to rebuild it, because it was so useful and nice to talk with people all over the planet. :-) I suppose you wouldn't be interested in a simple c program, that can be used from within a dialplan to let it ring and another hack to pick up the phone, without having a softphone? Kindly yours Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] meetme changes between asterisk 1.6.2.6 and 1.6.2.7
Hi Guys, is it possible that the silence joining with the Option q in a MeetMe room damaged ? I updated to Version 1.6.2.7 (before 1.6.2.6) and now my silence Orginates to Play Voice into a Meetme Room will play a bleep after a Success Orginate I Orginate with this simple AMI request. Action: Originate Channel: Local/1122 Application: Meetme Data: 1234,q, Can anyone reproduce this ? Thanx for your help. Daniel Knoll-- _ -- 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] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Fri, May 21, 2010 at 23:35:14 -0300, Tim Nelson wrote:
Greetings!
Hi, Tim!
I had the opportunity to test a Sangoma A200 card and I have some
doubts that I would like to consult:
I tried to detect the card and I had no success using the wctdm
module with DAHDI. I guess this is because electronics is different
because the TDM400P and OpenVox A400P cards have separate modules for
each channel, while the Sangoma A200 each module operates two
channels. I had to compile Wanpipe for the card was detected. Is it
the only way?
Wanpipe is what interfaces the hardware with Dahdi/Zaptel. Then,
Dahdi/Zaptel interfaces with Asterisk. This is normal.
Well, then wanpipe is necessary.
Another thing I want to try is to connect Asterisk with Siemens PBX
so that the extensions on Asterisk can communicate with the
extensions on the Siemens PBX and vice versa. For this should I
connect a FXO channel on Asterisk with a FXS channel of Siemens PBX?
Personally, if possible, I'd connect one of each(FXO/FXS) on Asterisk
to one of each(FXO/FXS) on the Siemens. This allows for proper dialing
between systems and passing your ${EXTEN} as expected.
I'm not sure if I understood well. Must I use two FXO/FXS connections? A
FXO (Asterisk) / FXS (Siemens) connection and another FXO (Siemens) /
FXS (Asterisk) connection? does not serve a single connection for
incoming and outgoing calls like when we connect Asterisk to the PSTN?
I noticed that, unlike OpenVox A400P card, RJ connectors on the
Sangoma A200 card are smaller. Apparently, the OpenVox use standard
telephone connectors.
Sangoma's cards come with a half-height PCI bracket for smaller
systems. To ensure the card stays small, they use smaller jacks, RJ14
or 'handset' jacks IIRC. Again, this is something specific to Sangoma
and normal.
Today I was doing tests connecting FXO channel on Sangoma card to a
extension of Siemens PBX. Previously, connecting a phone, I made sure in
that socket I had a dial tone.
I tried calling the extension 509 on Siemens PBX, but I get a busy tone
with the following message in the CLI:
- -
dynatac*CLI
-- Executing [9...@from-internal:1] Dial(SIP/200-0004,
DAHDI/3/509) in new stack
[May 26 14:47:59] WARNING[3031]: app_dial.c:1298 dial_exec_full: Unable
to create channel of type 'DAHDI' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9...@from-internal:2] Hangup(SIP/200-0004, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0004'
-- Executing [9...@from-internal:1] Dial(SIP/200-0005,
DAHDI/3/509) in new stack
[May 26 14:48:32] WARNING[3032]: app_dial.c:1298 dial_exec_full: Unable
to create channel of type 'DAHDI' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [9...@from-internal:2] Hangup(SIP/200-0005, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0005'
- -
This is the configuration I'm using in chan_dahdi.conf:
- -
[trunkgroups]
[channels]
language=es
defaultzone=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
inmediate=no
; DGB - 20100322
busydetect=yes
busycount=3
;Sangoma AFT-A200 [slot:8 bus:1 span:1] wanpipe1
context=from-internal
mailbox=...@voicemail
callerid=Jane Doe 300
group=1
echocancel=yes
signalling = fxo_ls
channel = 1
context=from-internal
group=2
echocancel=yes
signalling = fxo_ks
channel = 2
context=from-zaptel
group=3
echocancel=yes
signalling = fxs_ks
channel = 3
context=from-zaptel
group=4
echocancel=yes
signalling = fxs_ks
channel = 4
-
Re: [asterisk-users] Dahdi problems with kernel 2.6.32
Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, May 23, 2010 at 04:54:38AM -0400, cov...@ccs.covici.com wrote: Hi. I am having problems with dahdi using kernel 2.6.32. I am using 2.6.30 kernel and it works fine -- here is the output of dahdi_cfg -vv ++ dahdi_cfg -vv DAHDI Tools Version - SVN-trunk-r8670 DAHDI Version: SVN-trunk-r7445M Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 2 channels to configure. Changing signalling on channel 1 from Unused to FXO Kewlstart Setting echocan for channel 1 to mg2 Changing signalling on channel 4 from Unused to FXS Kewlstart Setting echocan for channel 4 to mg2 However, if I am using kernel 2.6.32 I get the following -- the configs have not changed: ++ dahdi_cfg -vv DAHDI Tools Version - SVN-trunk-r7409 DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) Any chance that with that version the device was not created? What do you see on the output of lsdahdi in both cases? DAHDI Version: SVN-trunk-r8653M Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 2 channels to configure. I had to update dahdi-trunk because the old version would not compile under 2.6.32, and update of dahdi-tools-trunk make no difference. If you want a more stable version that still builds with newer kernels, look at the 2.2 or 2.3 svn branches (and released tarballs). From another thread, I blacklisted netjet and now things are working. But I wonder what is going on here and where did netjet come from -- it doesn't look like an dahdi module to me. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- 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] q931.c modifications for CLID Presentation
Anyone can comment on this please? Is it right to assume that if you own a PRI Caller ID always comes through even if customer used *67 feature to block their CLID? I understand that is true of calling a Toll-Free number. Does Asterisk or LibPRI somewhere in the code abide by some standard to strip or hide the CLID if Callee requested private presentation? Thanks On Sat, May 15, 2010 at 4:14 PM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, We have a problem with Caller ID not being displayed. I want to test everything to see where the problem is with the incoming Caller ID and why it's not displaying. I am tracking this down to Presentation prohibited of network provided number even though the Caller doesn't use *67 and even though they haven't asked their provider to block their CLID for outbound. I want to make a modification to q931.c or pri_facility (whichever responsible) to ignore the request from the network to prohibit CLID and to show it so that I can find out exactly where the problem lies. Can you please tell me which if is related to that in q931.c or pri_facility.c? 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
[asterisk-users] BRI card(B800P) doesn's work with DAHDI(wcb4xxp) in NT mode
Dear Supports, I was attempting to install BRI Card(OpenVox B800P) with wcb4xxp in NT mode .But I can not make it worked! Could you please give me some hints? Thanks in advance! Here are my environments: -- CentOS-5.3 Kernel-2.6.18-164.el5 asterisk-1.6.1.6 dahdi-linux-complete-2.3.0+2.3.0 libpri-1.4.10.2 OpenVox-B800P(the first port is spliting into two lines, I set the first one into NT mode,and others are TE mode) patch- manual_te_nt.patch(which download from:http://www.openvox.cn/bbs/viewthread.php?tid=992, and posted by Dmitry) - I make a fresh OS and yum install all the dependent packages, like bison,bison-devel,gnults-devel,gcc,gcc-c++, and so on. And download all the same versions of DAHDI-linux-complete-2.3.0+2.3.0, asterisk-1.6.1.6, libpri-1.4.10.2 from asterisk official website:http://downloads.asterisk.org/pub/telephony/. Completely plugs the card and start to compile and install those packages. Here are my installation steps: A.Libpri: make--make install --(no errors) B.Patch file /usr/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/wcb4xxp/base.c with this method: patch p0 manual_te_nt.patch C.DAHDI: make--make install--make config --(no errors) D.Edit and insert this line at the buttom of file /etc/modprobe.d/dahdi.conf(only this step is different from yours) options wcb4xxp te_nt_override=0xFE E.Start dahdi and automatically configure the channels with: modprobe dahdi--modprobe wcb4xxp debug=16--dahdi_scan--dahdi_genconf--dahdi_cfg -v (no errors and all the channels are showing there) [r...@localhost modprobe.d]# dahdi_cfg - DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 5: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 6: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 7: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 8: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: none) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: none) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: none) (Slaves: 05) Channel 06: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: none) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: none) (Slaves: 08) Channel 09: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: none) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: none) (Slaves: 11) Channel 12: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: none) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: none) (Slaves: 14) Channel 15: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 15) Channel 16: Clear channel (Default) (Echo Canceler: none) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: none) (Slaves: 17) Channel 18: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: none) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: none) (Slaves: 20) Channel 21: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: none) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: none) (Slaves: 23) Channel 24: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 24) 24 channels to configure. Setting echocan for channel 1 to none Setting echocan for channel 2 to none Setting echocan for channel 3 to none Setting echocan for channel 4 to none Setting echocan for channel 5 to none Setting echocan for channel 6 to none Setting echocan for channel 7 to none Setting echocan for channel 8 to none Setting echocan for channel 9 to none Setting echocan for channel 10 to none Setting echocan for channel 11 to none Setting echocan for channel 12 to none Setting echocan for channel 13 to none Setting echocan for channel 14 to none Setting echocan for channel 15 to none Setting echocan for channel 16 to none Setting echocan for channel 17 to none Setting echocan for channel 18 to none Setting echocan for channel 19 to none Setting echocan for channel 20
