Re: [asterisk-users] alwaysauthreject=yes not working as expected
Asterisk 1.4.42 Set alwaysauthreject=yes in [general] section of sip.conf. Restarted asterisk However when I attempt to register I still get: [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from 'sip:000333082261...@domain.com' failed for '121.98.1.1' - Wrong password [2012-08-08 21:12:42] NOTICE[15689] chan_sip.c: Registration from 'sip:00033308226...@domain.com' failed for '121.98.1.1' - No matching peer found Based on the Asterisk security advisory (http://downloads.asterisk.org/pub/security/AST-2011-011.html) I would have expected 1.4.42 to respond the same in both cases (since the issue was fixed in 1.4.41.2). Am I missing something obvious? Yes. Those are log messages for the administrator's benefit. They are not SIP messages sent in response to the REGISTER request. The SIP messages sent are supposed to be the same not the logging messages. Yes I agree they are supposed to be the same but they are not. Below is the dialog when a wrong password is provided with alwaysauthreject=yes: U 121.98.1.1:1025 - 203.89.1.1:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-d88996fba8b1fd8c-1---d8754z- ;rport..Max-Forwards: 70..C ontact: sip:1232261336@192.168.1.103:5060;rinstance=da68419a02006162. .To: sip:1232261...@domain.com..From: sip:123 2261...@domain.com;tag=f910aa53..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 1 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANC EL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 5.0.0 stamp 67284..Content-Length: 0 U 203.89.1.1:5060 - 121.98.1.1:1025 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-d88996fba8b1fd8c-1---d8754z- ;received=121.98.1.1;rport=1025..From: sip:000333 082261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY 2E...CSeq: 1 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0 U 203.89.1.1:5060 - 121.98.1.1:1025 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-d88996fba8b1fd8c-1---d8754z- ;received=121.98.1.1;rport=1025..From: sip: 1232261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com;tag=as16fea110..Call- ID: ZmM4YTU4NTg2MWNhYzVk YTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 1 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: repla ces..WWW-Authenticate: Digest algorithm=MD5, realm=domain.com, nonce=2f48b121..Content-Length: 0 U 121.98.1.1:1025 - 203.89.1.1:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-5c88940128ede618-1---d8754z- ;rport..Max-Forwards: 70..C ontact: sip:1232261336@192.168.1.103:5060;rinstance=da68419a02006162. .To: sip:1232261...@domain.com..From: sip:123 2261...@domain.com;tag=f910aa53..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 2 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANC EL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 5.0.0 stamp 67284..Authorization: Digest username=1232261336,re alm=domain.com,nonce=2f48b121,uri=sip:c-vm- 02.domain.com,response=cb74a7805412a3ac198800aeede3c06e,algorit hm=MD5..Content-Length: 0 U 203.89.1.1:5060 - 121.98.1.1:1025 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-5c88940128ede618-1---d8754z- ;received=121.98.1.1;rport=1025..From: sip:000333 082261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY 2E...CSeq: 2 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0 SIP/2.0 403 Forbidden (Bad auth)..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-5c88940128ede618-1---d8754z- ;received=121.98.1.1;rport=1025..Fro m: sip:1232261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com;tag=as16fea110..Call-ID: ZmM4YTU4NTg2 MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 2 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supporte d: replaces..Content-Length: 0 Is this a bug or am I missing something obvious? -- _ -- 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] alwaysauthreject=yes not working as expected
Asterisk 1.4.42 Set alwaysauthreject=yes in [general] section of sip.conf. Restarted asterisk However when I attempt to register I still get: [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from 'sip:000333082261...@domain.com' failed for '121.98.1.1' - Wrong password [2012-08-08 21:12:42] NOTICE[15689] chan_sip.c: Registration from 'sip:00033308226...@domain.com' failed for '121.98.1.1' - No matching peer found Based on the Asterisk security advisory (http://downloads.asterisk.org/pub/security/AST-2011-011.html) I would have expected 1.4.42 to respond the same in both cases (since the issue was fixed in 1.4.41.2). Am I missing something obvious? -- _ -- 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] Block outbound calls based on IP address
Thanks for the reply however it is not possible to get the public IP address using the SIP_HEADER function (see my original post). We have many devices connecting from hundreds of dynamic external IPs. -- _ -- 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] Block outbound calls based on IP address
Thanks.
exten = s,n,Set(foo=${CHANNEL(peerip)}) ; Doesn't return anything
exten = s,n,Set(foo=${CHANNEL(recvip)}) ; Doesn't return anything
exten = s,n,Set(foo=${SIPCHANINFO(peerip)}) ; Returns public IP when
calling from a SIP device
Strange that CHANNEL doesn't return anything.
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[asterisk-users] Block outbound calls based on IP address
We are looking to further secure our Asterisk installation by inspecting the IP address that a SIP INVITE comes from and performing some logic to determine whether the call should proceed. The purpose of this is to prevent calls to certain expensive destinations if the SIP message is coming from a foreign IP that we don't expect. I can see that it's possible to use the SIP_HEADER function however that may not contain the public IP address. For example here is an invite from the external IP address 58.28.1.1 but that information is not contained in the SIP header: U 58.28.1.1:5060 - 203.89.1.1:5060 INVITE sip:1...@domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-fc116e03a80ef774-1---d8754z-;rport. .Max-Forwards: 70 ..Contact: sip:000333082261336@192.168.1.103:5060..To: sip:1...@domain.com..From: sip:000333082261...@domain.com;tag=7 dcb1e4d..Call-ID: NDMyZmRhY2Q4ZjNhMjAxMDJhOTA3OTU0MzMyNTkzNjI...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O..Content-Type: application/sdp..Supported: replaces..User-Agent: X-Lite release 5.0.0 stamp 67284..Content-Length: 217v=0..o=- 12988751314362048 1 IN IP4 192.168.1.103..s=CounterPath X-Lite 5.0.0..c=IN IP4 192.168.1.103..b=AS:1638..t=0 0..m=audio 5062 RTP/AVP 0 8 3 101..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-15..a=sendrecv.. Is it possible to determine the public IP address from the dialplan? Any advice appreciated. attachment: winmail.dat-- _ -- 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] No progress tones on transferred call
Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2) to 1595 (SPA922 000B820AF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4A). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the extension is dialled: U 203.89.001.001:5060 - 121.98.001.001:1034 INVITE sip:1CDF0F4A@192.168.1.72:5060 SIP/2.0..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e;rport..From: C Allerid sip:1593@203.89.001.001;tag=as72616c50..To: sip:1CDF0F4A@192.168.1.72:5060..Contact: sip:1593@203.89.001.001..Call-ID: 59ba10300b9b8cb5684eba2368c90...@203.89.001.001..cseq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 05 Jun 2012 08:05:02 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Type: application/sdp..Content-Length: 262v=0..o=root 3031 3031 IN IP4 203.89.001.001..s=session..c=IN IP4 203.89.001.001..t=0 0..m=audio 13728 RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:1010-16..a=ptime:20..a=sendrecv.. U 121.98.001.001:1034 - 203.89.001.001:5060 SIP/2.0 100 Trying..To: sip:1CDF0F4A@192.168.1.72:5060..From: C Allerid sip:1593@203.89.001.001;tag=as72616c50..Call-ID: 59ba10300b9b8cb5684eba2368c90...@203.89.001.001..cseq: 102 INVITE..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Server: Cisco/SPA508G-7.4.9c..Content-Length: 0 U 121.98.001.001:1034 - 203.89.001.001:5060 SIP/2.0 180 Ringing..To: sip:1CDF0F4A@192.168.1.72:5060;tag=53e23c5265d60f06i0..From: C Allerid sip:1593@203.89.001.001;tag=as72616c50..Call-ID:59ba10300b9b8cb5684eba2368 c90...@203.89.001.001..cseq: 102 INVITE..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Contact: $USER sip:1CDF0F4A@192.168.1.72:5060..Server: Cisco/SPA508G-7.4.9c..Content-Length: 0 After transfer is pressed the second time there is no further SIP messages with Asterisk CLI -- Executing [s@macro-dial:12] Dial(SIP/000B820AF-2d0a, SIP/000E08D6SIP/1CDF0F4ASIP/000E08D61|20|tTwWr) in new stack -- Called 1CDF0F4A -- SIP/1CDF0F4A-2d0b is ringing -- Stopped music on hold on SIP/0026998D2-2d08 Updated sip.conf progressinband=yes This didn't make any difference I've tried calls in different directions in case it is to do with the particular phone firmware but the direction is irrelevant. Any suggestions appreciated or if you require further information please ask. -- _ -- 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] No call progress sounds
On Nov 8, 2011, at 9:55 AM, isr...@gmail.com wrote: There is a bug which blocks call progress message 8 which was fixed but I don't remember in which version Try upgrading to latest 1.6 version Before we opened for the day today I updated to 1.6.2.20 and that seems to have solved the call progress problem. Thanks -chris www.mythtech.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No call progress sounds
I recently switched to a PRI from analog lines. For reasons out of my control, my vendor had problems getting the PRI to interface so they set it to use T1-CAS instead. The lines are working just fine for inbound and outbound calls, except I get no call progress sounds. So no ring, busy, etc. When you place an outbound call, you just have dead air until the called party picks up. If it is a busy number, you have no way to know as it just sits with dead air until you give up and hang up. I have two lines for faxing stripped out of the T1 from the router and those have all proper audio, so this may very well be something misconfigured on my end, but I can't figure out what. I am using a Sangoma A101 card for the interface and running Asterisk 1.6.2.13. Anyone have any idea what I need to change to get the call progress audio? Thanks! -chris www.mythtech.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISAC and Asterisk
On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote: Are there any plans to include the ISAC codec in Asterisk? Is it possible or even desirable? Is ISAC open source (nothing indicates it is from the WebRTC website http://www.webrtc.org)? What do you need it for? The possibility of having a web-based softphone without requiring any plug-in is interesting. The adaptive nature of the ISAC codec could also prove useful. I see lots of possibilities in the mobile device space. I guess the lack of responses gives me the answer anyway! -- _ -- 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] 64 pickup groups
We have multiple customers running on a single Asterisk 1.4 installation and therefore require a large number of pickup groups. There seems to be a limitation of 64 call groups. Can anyone suggest how we work around this? For example is this limitation removed in a later version, is there a patch, are we approaching this in the wrong way? Any advice appreciated. attachment: winmail.dat-- _ -- 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] ISAC and Asterisk
Are there any plans to include the ISAC codec in Asterisk? Is it possible or even desirable? Is ISAC open source (nothing indicates it is from the WebRTC website http://www.webrtc.org)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Invalid use of undefined type when make dahdi
I am attempting to install Dahdi on a virtual machine running Centos 5.5 and
having various problems.
yum install kernel-devel gcc make gcc-c++ libxml2-devel
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
* base: mirror.optus.net
* extras: mirror.optus.net
* rpmforge: fr2.rpmfind.net
* updates: mirror.optus.net
Setting up Install Process
Package kernel-devel-2.6.18-238.9.1.el5.x86_64 already installed and
latest version
Package gcc-4.1.2-50.el5.x86_64 already installed and latest version
Package 1:make-3.81-3.el5.x86_64 already installed and latest version
Package gcc-c++-4.1.2-50.el5.x86_64 already installed and latest version
Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.x86_64 already installed
and latest version
Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.i386 already installed and
latest version
Nothing to do
[root@atlantis dahdi-linux-2.4.1.2]# make
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
make[1]: Leaving directory
`/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
You do not appear to have the sources for the 2.6.18-238.el5 kernel
installed.
So we're running a different kernel...
uname -r
2.6.18-238.el5
Downloaded kernel sources for 2.6.18-238.el5
[user@atlantis ~]$ mkdir -p ~/rpmbuild/{BUILD,RPMS,SOURCES,SPECS,SRPMS}
[user@atlantis ~]$ echo '%_topdir %(echo $HOME)/rpmbuild'
~/.rpmmacros
[user@atlantis ~]$cd ~/rpmbuild/SPECS
[user@atlantis ~]$rpmbuild -bp --target=`uname -m` kernel-2.6.spec
2 prep-err.log | tee prep-out.log
[root@atlantis kernel-2.6.18]# cp
/home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
/usr/src/kernels/2.6.18-238.el5-x86_64 -R
[root@atlantis dahdi-linux-complete-2.4.1.2+2.4.1]# make all
/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
error: invalid use of undefined type ‘struct module’
/usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
error: ‘struct dahdi_chan’ has no member named ‘pulsecount’
etc etc
This article
(http://asteriskfaqs.org/2011/01/30/asterisk-users/invalid-use-of-undefined-type-struct-module.html)
indicates that those errors are the result of not having CONFIG_MODULES set in
the kernel config.
cd /home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
make menuconfig
[*] Enable loadable module support
Legend: [*] built-in
Any advice appreciated.
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Re: [asterisk-users] Invalid use of undefined type when make dahdi
On Wed, May 04, 2011 at 09:56:40PM +1200, CB wrote:
I am attempting to install Dahdi on a virtual machine running Centos
5.5 and
having various problems.
yum install kernel-devel gcc make gcc-c++ libxml2-devel
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
* base: mirror.optus.net
* extras: mirror.optus.net
* rpmforge: fr2.rpmfind.net
* updates: mirror.optus.net
Setting up Install Process
Package kernel-devel-2.6.18-238.9.1.el5.x86_64 already installed and
latest version
Package gcc-4.1.2-50.el5.x86_64 already installed and latest version
Package 1:make-3.81-3.el5.x86_64 already installed and latest version
Package gcc-c++-4.1.2-50.el5.x86_64 already installed and latest
version
Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.x86_64 already installed
and latest version
Package libxml2-devel-2.6.26-2.1.2.8.el5_5.1.i386 already installed
and
latest version
Nothing to do
[root@atlantis dahdi-linux-2.4.1.2]# make
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
make[1]: Leaving directory
`/usr/src/dahdi-linux-2.4.1.2/drivers/dahdi/firmware'
You do not appear to have the sources for the 2.6.18-238.el5 kernel
installed.
So we're running a different kernel...
uname -r
2.6.18-238.el5
Downloaded kernel sources for 2.6.18-238.el5
[user@atlantis ~]$ mkdir -p
~/rpmbuild/{BUILD,RPMS,SOURCES,SPECS,SRPMS}
[user@atlantis ~]$ echo '%_topdir %(echo $HOME)/rpmbuild'
~/.rpmmacros
[user@atlantis ~]$cd ~/rpmbuild/SPECS
[user@atlantis ~]$rpmbuild -bp --target=`uname -m` kernel-2.6.spec
2 prep-err.log | tee prep-out.log
[root@atlantis kernel-2.6.18]# cp
/home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
/usr/src/kernels/2.6.18-238.el5-x86_64 -R
[root@atlantis dahdi-linux-complete-2.4.1.2+2.4.1]# make all
/usr/src/dahdi-linux-complete-
2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
error: invalid use of undefined type ‘struct module’
/usr/src/dahdi-linux-complete-
2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652:
error: ‘struct dahdi_chan’ has no member named ‘pulsecount’
etc etc
This article (http://asteriskfaqs.org/2011/01/30/asterisk-
users/invalid-use-of-undefined-type-struct-module.html) indicates that
those errors are the result of not having CONFIG_MODULES set in the
kernel config.
cd /home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64
make menuconfig
[*] Enable loadable module support
Legend: [*] built-in
Any advice appreciated.
It still looks like the kernel that is running doesn't match the kernel
that
you prepped.
Can you yum install kernel-devel-`uname -r` after cleaning up the
/usr/src/kernels/2.6.18-238.el5-x86_64 directory and then build?
mv 2.6.18-238.el5-x86_64 /tmp
yum install kernel-devel-`uname -r`
make clean
make all
make install
service dahdi start
Loading DAHDI hardware modules:
No hardware timing source found in /proc/dahdi, loading dahdi_dummy
Running dahdi_cfg: [ OK ]
Nice!
Thanks very much.
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Re: [asterisk-users] sangoma card rx/tx gain level
On Apr 15, 2011, at 12:50 PM, satish patel wrote: We had echo issue before so we replaced old PRI card with Sangoma A102D HWEC. Now my question is i set rx/txgain level 0.0 default do i need to touch this value or default is best. I have read on google and people say it should around 14844 on ztmonitor for rx/tx level same. I just use milliwatt and test my default 0.0 rx/tx level and it come around 4600. Do you think i need to make it around 14844 ? I have some Sangoma cards I've been running for several years at two different locations and never had to touch the gain levels. They run beautifully at the default of 0.0. I also have some running at one location that I had to change the levels quite a bit. In that instance, I found the recommended 14844 was off by about a factor of 10. If I tried to get those levels, things were obviously way out of line, all sorts of distortion would happen. I ended up using ztmonitor, but I just watched the gauge and tuned it like audio recording equipment. I targeted about 2/3 to 3/4 up the graph as the average level. I didn't find a reliable milliwatt number to test on, so I just watched a bunch of normal calls. Once I had things working at the best level testing by ear and watching the graph, I found the value ztmonitor reported was about 4500-5000, or about 10 times lower than the info I found via google (I'm guessing I saw the same page you saw with the 14844 value). -chris www.mythtech.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 and Realtime
Could anyone recommend some documentation regarding Asterisk 1.8 and the realtime architecture? Specifically I want to know if it is possible to set a priority label or to use n as a priority for realtime extensions in Asterisk 1.8? My understanding is that is not possible with Asterisk 1.4 and I wonder if it's changed? attachment: winmail.dat-- _ -- 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] Echo on Sangoma A400 and background noise
On Sep 15, 2010, at 6:10 PM, Al lists wrote: we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and oslec echo canceler. I've only used Sangoma with hardware echo cancelation, but I did find I had to manually tune my cards. In my case using a test tone number was of no help, rather I just listened in on calls in progress and used ztmonitor to see where the levels were and adjusted things until I had a good level and the sound quality was acceptable. Until I did the manual tuning I had serious issues of static and background noise on several lines, but not all. I don't blame this on the Sangoma card, but rather on the line quality as I run Sangomas in other locations and have never had to do any tuning there, they have just worked. -chris www.mythtech.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones won't stop ringing
On Mar 10, 2010, at 10:27 PM, Chris Owen wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? All Aastra 6755i I've been seeing this lately on Cisco 7940, seems to happen on two of the three at a location I deal with. They worked fine for years and then all of a sudden this just started happening. Rebooting the phone will cure it for a period of time, but it always comes back, and always to the same two phones (although not always at the same time). I don't think anything changed when it started happening, but I can't say for sure. It may also happen on a Polycom at that location as well, reports on that one have been sketchy, so I can't be sure it really is versus they are hearing a 2nd call ringing and just think the phone is stuck ringing. (I do know for a fact it happens with the Cisco and is not simply a 2nd call). I had figured it was the old version of Asterisk I'm running and the fact that the server has had several power failures so who knows the health of the machine and install. But if it is happening to others, my assumption may be wrong. -chris www.mythtech.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
On Jan 24, 2010, at 12:22 PM, Karl Fife wrote: --I can adjust the volume easily without looking AND without fat- fingering some DTMF tones--very good haptics. With the Snom you have to look and guide your fingers to the volume buttons or you'll inadvertently beep some DTMF's. Dumb. So too the MUTE button. I use the Snom 370 all day long at work. I have never had a problem adjusting the volume. I change it multiple times a day as I keep my handset on one volume and my headset on another, so I'm always going up and down and I've never accidentally pressed any other key. I will however agree with you on the Mute button, any time I want to mute a call, I have to stop and look at the buttons and figure out which one of the tiny ones is mute. Maybe that is because I don't mute very often, but it is still a very small button mixed in with other small buttons. However, there is probably nothing stopping me from programming one of the 12 programmable buttons as a Mute button giving me a larger more convenient target. I've never tried as like I said, I don't use mute very often. --With the Snom it's very easy to leave DND ON accidentally. The indicator is tiny. Imaging leaving yoru phone on DND all morning :-). I have done this several times. I've found the DND, being as it is on the bottom corner of a row of buttons, is easy to accidentally hit if your hand hits the phone when reaching for something else or when going to hit another button. And like you said, it is a tiny symbol on the display and not easy to notice. Usually I figure it out when I someone specifically tells me they are transferring a call to me and then it doesn't come thru. I average probably about one a month that I do this. I know others have done it as well as they call me when their phone isn't receiving any inbound calls and when I check sure enough, they too have turned on DND. Again, the Snom allows pretty liberal programming so it would likely be possibly to disable that button from doing DND and change to something else less likely to accidentally be hit. However that won't resolve the issue of the display icon being so small allowing you to easily forget you turned it on in the first place. --Another DUMB charactaristic of the Snom is that you can't 'hang up' NOR change SIP registrations without actually placing the receiver back in the cradle. WTF? I've never tried to change registrations with the handset off hook, but I do hang up the phone all the time without putting the handset back. Press the X button, it will hang up the phone regardless of handset hook state. When you are ready to make another call, the Check button will take the phone back off hook (or in the case of how mine are set up, I just dial the next number and when it matches my dial plan it goes off hook automatically and dials). This is also how I take the phone and and off hook for my headset or for speaker phone. I leave the handset on hook and use the Check and X buttons to toggle the hook state to answer and hang up calls. Furthermore if you 'end call' it will play dial tone again immediately, and there's no way to shut it up without replacing the handset. I'm guessing you are talking about using the X button here. Weird that yours plays the dial tone again right away as mine does not. If I press X with the handset off hook it hangs up the phone and leaves it hung up until I either press Check or manually depress and release the handset hook switch (or dial something matching my dial plan or anything else that should specifically trigger the phone going off hook). I wonder if this could be a difference in firmware versions. I believe I am currently running 7.3.24 although I could be wrong (but it should be one around there). One thing that does drive me nuts about my Snom, if I am on a call, and have another ringing, I can't change to a different talk method (handset/headset/speaker) until the ringing call on call waiting stops. If I try, it will automatically put the first call on hold and answer the 2nd call. I've had this happen several times as I change from handset to headset mid call and find myself suddenly talking to a different caller. -chris www.mythtech.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about static
On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote: Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc….. Typically a reboot of the phone resolves the problem…person also swears there is nothing on or near their desk to cause interference (microwave, cell phone is purse). Only one user? Did you check to see if it is a bad handset cord? -chris www.mythtech.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Clarifying RX and TX gains
I've done gain tuning as per the info I've found online. I've got my RXGain set so my volumes list as about 14,800 (using a milliwatt test number and ztmonitor -vv). However listening to the line now, this sounds too loud to me. The person speaking sounds fine, but I've now got a large amount of background hiss coming thru. In order to get the recommended levels, most of my lines are in the range of +12.5, so I'm wondering if I'm just exceeding the ability of my Sangoma card to amplify the audio. My TXGain's are all still at 0.0 as the Sangoma card didn't seem to want to let me adjust the gains. My levels are listing in the 5,000 range, but it seemed no matter what I changed the values to, there was virtually no difference made to the audio levels. Listening to the transmit levels everything sounds fine. What I am still getting is occasional static on the lines. Pops, squeaks, that kind of stuff. Sounds exactly like a handset cord is going bad, but it happens on any phone (and all have brand new handset cords). The person on the far end says they hear nothing wrong with the call. However, when I have asterisk record the call, the static is present on the recording. I also get the occasional robotic voice echoing back on anything said by the local person. That sounds like it should be typical echo on the line (I'm using a Sangoma A400DE with hardware echo cancellation). Everything I can find on echo and static tells me I should adjust my RX and TX gains. So that brings me to my confusion/clarification. I've got my RXGain set to the recommended level, and my TXGain doesn't seem to be very adjustable upwards as it would appear to need based on the recommended levels. Could my static and echo be caused by my TXGain being too low, or possibly RX or TX actually being too high (since listening to the RX reveals lots of background hiss)? I'm trying to work out this last little bit on my system as I'm getting grief that these calls have audio issues. I'm running out of things to check, so I'm looking for clarification on where exactly my RX and TX gains should be and which directions they will impact static and echo. Thanks! -chris www.mythtech.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio distorted local side only
I'm not sure where to check next, so I'm reaching out to those that know this stuff better than I. I've got Asterisk up and running, but I've still got an occasional audio issue. Once in a while (maybe 1 out of every 20-30 calls), the audio becomes heavily distorted, but only on the local side. The party on the other end says the audio is fine. We can hear them, although on these calls the remote party is quieter than usual. They can hear us just fine. But when we speak, our side becomes heavily distorted. It only happens when we speak. As long as we are not talking, the audio is fine other than the remote party is not as loud as normal. The distortion sounds a lot like what a speaker sounds like when it is blown. Not quite static, not quite feedback, but very crackly and metallic with pops and squeals. It drowns out all other sound on the call. We hear it very loud, almost painfully and you have to hold the handset away from your ear as you talk. The remote party says they hear nothing at all wrong. It happens on both handset and speaker phone, so it shouldn't be a speaker issue. This never seems to happen on interoffice calls, only on calls going out over our POTS lines. Also, if we put the call on hold or transfer the call, usually the distortion is gone once the call is picked back up (although it may come back, but not as bad). Flip side, sometimes a good call that gets put on hold or transfered will have the issue when the call is picked up again. I've done rxgain tuning, although I can't be 100% sure I've done it right. I followed the advice from http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/ , but the levels he said to aim for cause me to need to raise my lines to between +12 and +20 and when listening to the tone on a phone sound as if they are starting to clip (although that could be correct for all I know). I had been doing txgain tuning, but in my testing, I found that either the values I was setting made no difference in the audio levels or topped out the audio levels around 9000 instead of the article's recommended 14844. As such, my txgain is currently set to 0 for all lines. Also it is worth noting that I was not able to turn up a milliwatt test number for my local CO (I even asked a Verizon tech who claimed they didn't have one in my CO and he didn't know the number for one in the area, although he may have just been BSing me to not give it out). The number I used is several states away, so I don't know how much (if any) audio loss is present on the tone, thus it is possible my rxgain may be too high for my lines. If anyone knows of a milliwatt test number for northern NJ I'd be happy to repeat my tuning. I am using Snom 370 phones (running the latest 7.3.23 firmware, but it was happening with the 7.3.14 firmware as well). My Analog lines are connected via a Sangoma A400DE. This is their 12 port card with hardware echo cancel and PCI-E. The network is a dedicated network, only the phones are on it, and all are connected to a netgear 10/100/1000 POE switch. The phones are using POE. The only devices on the network that are not phones are the Asterisk server, a router to get the network access to the internet (currently outbound only, I've not yet allowed inbound access to the server) and my laptop for configuring the server (although the issue happens even without my laptop connected). So this likely is not caused by traffic killing the network (won't rule it out, but it doesn't seem like a first guess). Anyone know what is going on or can tell me where to go next to figure it out? Thanks for any -chris www.mythtech.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma FXS dialmap
On May 13, 2009, at 10:50 AM, Doug Lytle wrote: I have a Sangoma A400 card with two FXS ports. They work fine, however as I have analog phones connected, I have no way of telling the phone I am done dialing. Pressing # works fine, but then Asterisk That's what the digit and response timeouts are for. I have: ; ; Set Timeouts ; exten = s,n,Set(TIMEOUT(response)=8) exten = s,n,Set(TIMEOUT(digit)=2) I'm sorry to be really thick here, but where exactly would I place these lines. This is probably due to my misunderstanding of the sequence of things, so it may be obvious and I'm just not getting it. I have my FXS extensions defined in my zapata.conf file, but that doesn't seem like the right place to put what looks like dialplan commands. My sip phones are likewise all configured in sip.conf (and not really an issue here as those have their own dialmaps and handle all this without a problem). The above lines look like they belong in the extensions.conf, but I thought that was only called once a phone sent the dial string to asterisk. Thus my confusion of setting timeout values for a phone connected to an FXS port in an area that I though it didn't look at until after the FXS port sent the dialing (and naturally I'm trying to get the phone/FXS port to send the dialing). Again, this is probably due to my misunderstanding of how, when, and where things are called, so the answer as to where to put the above commands may be obvious to others. But I didn't want to just take a guess and risk screwing things up. Where is the correct place to put those two commands so my analog phones connected to the FXS ports will use them to decide when dialing has completed. Thanks again! -chris www.mythtech.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma FXS dialmap
On May 14, 2009, at 11:34 AM, Doug Lytle wrote: SIP phones send a completed dial string, analog phones send 1 digit at a time. With the timeout values, it no more digits are recieved by the 2 second timeout, the dial plan continues. Ok, that was the part I didn't understand that makes it all make sense now. I thought analog (FXS) ports held the dialing until told it was complete as well. Now knowing they send one digit at a time makes it all clear. Thanks for the help! -chris www.mythtech.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma FXS dialmap
I have a Sangoma A400 card with two FXS ports. They work fine, however as I have analog phones connected, I have no way of telling the phone I am done dialing. Pressing # works fine, but then Asterisk passes that # over to the POTS line, and about every 5th call, for some reason that is causing the call on the POTS line to fail. The suspect the trailing # is also going to get in the way of transferring to another extension (or any other dial codes that might need to be used from these analog phones). Is their either a way to strip the # off the end of the dialing so it isn't passed thru, or is there a place I can specify a dialmap for the FXS ports so when my dial pattern is matched it just dials right away (or set a dial timeout as right now there doesn't appear to be one, it just waits forever for you to finish dialing). Or is none of this possible with FXS ports on Sangoma cards and I should look at getting an external ATA where I can set these things. -chris www.mythtech.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
On Oct 29, 2007, at 5:35 PM, Kyle Sexton wrote: Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg I don't know who makes the above phone, but physically, it looks nearly identical to the SBC 125 or SBC 225 http:// www.sbcphonestore.com/SBC-Corded-Telephones/1-Line-Multifunction- Caller-ID-Speakerphone-SBC-125-ii_2.html I have no idea if SBC makes their phone themselves or contract it out to someone else. But going just off look, I'd think the SBC phone and your mystery phone clearly have some part of the manufacturing process in common, because it is definitely using the same shell. I have access to a few of the SBC 120 (also the same case, but lacks the little side panel for speed dial info), so if you really need to know more, I can look for FCC numbers or other info to try to determine who the ultimate manufacturer is for the SBC phone. -chris www.mythtech.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set up two PSTN calls and then join them
I wish to set up two PSTN calls and then connect them similar to Jajah (is this called 3pcc?). The PSTN interconnect is handled by a third party SIP provider. I can do this using the manager or call files. An example (using php) would be: fputs($oSocket, Action: login\r\n); fputs($oSocket, Events: off\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Exten: $strExten\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Priority: $strPriority\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); This simulates one extension calling another and has some limitations e.g. the CDR records show one call which doesn't meet my requirements for billing purposes. What's the best way to achieve this join up two calls scenario? Any suggestions appreciated. Cameron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
On Sep 26, 2007, at 10:58 AM, Ricardo Carvalho wrote: All phones have firmware version 1.1.1.14; we are testing new stable version 1.1.4.18 but by now we found that some phones freeze sometimes - version 1.1.1.14 seems more stable. I'm not sure which firmware I'm running on my GXP2000 (I only have one at current), but I did find that it does not like my DHCP server. If I set the phone to use DHCP it would freeze periodically when it tried to renew the DHCP lease. I'd have to yank the power to get it to reset. Changing to static IP fixed the problem and I haven't had any freezing with it since (6 months+ now). I can't say if this is specifically a Grandstream issue with the DHCP as I also know that Windows 98 doesn't like my DHCP server either and fails to renew leases properly as well. So I could just have a crappy DHCP server in place at that location and it may be the true source of the Grandstream's DHCP problems. Figured I'd let you know in case you are using DHCP you may want to try static and see if the freezing stops for you as well. -chris www.mythtech.net ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to stack Sangoma Remora cards
On Aug 7, 2007, at 1:28 AM, Olivier wrote: How can can you stack sangoma cards such as http://www.sangoma.com/ datasheets/p_a200-specs in a given PC enclosure ? It seems to me that it introduces mechanical constraints that seem difficult to comply with as space between cards is set by Remora expansion card. The space between the cards (as dictated by the Remora expansion card), works out to keep the additional stacked cards in the correct spacing for standard PCI or ISA slot distances. So basically, each stacked card you install, takes up the next slot over in your PC's case. So although they don't actually use the slot itself (minimizing IRQ and similar issues), they do occupy the physical space of the slot they hover over (and use that slot's opening to give access to the line jacks). If you have other expansion cards already in your PC that use the slot space, then you would not be able to use the Sangoma Remora expansion setup (unless you can rearrange the other cards to allow for enough consecutive open slots for the Remora backed cards). Hopefully that is the info you were looking for. -chris www.mythtech.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
On Jul 4, 2007, at 9:59 AM, Steve Kennedy wrote: Oh, so anyway, who was guy Eng you named the country after? And who was America named after ? Amerigo Vespucci -chris www.mythtech.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco remote reboot
On May 27, 2007, at 5:29 PM, Paul Aviles wrote: Is there a way to remote reboot a Cisco 7940 or 7960 phone via some kind of command? The idea is to force a reboot automatically after changing one of the configuration files. As long as you have telnet access turned on in the config file, you can telnet to them and issue a reboot from there. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?
On May 9, 2007, at 3:45 PM, Gavin Henry wrote: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci- express-p-393.html But it will be 3 PCI slots. Just to clarify in case you didn't already realize it. It doesn't actually *use* 3 PCI slots, it just occupies the physical space of 3. The board only connects to one slot, then has its own backplane that the additional daughter cards sit on. An important distinction if your concern with the use of 3 slots wasn't due to physical space, but rather was with dealing with IRQ and timing issues of having multiple slots in use. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
On Apr 18, 2007, at 6:50 AM, Rob Hillis wrote: We've had the very occasional problem with the phone locking up, but nothing overly serious. Are you using DHCP on the GXPs that are locking up? I have one and it would lock up almost every night requiring the power to be pulled in the morning. Knowing my DHCP server can sometimes be a PITA and not renew leases properly, I on a hunch changed my GXP to a static IP address and so far it has yet to lock up again. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudspeaker
On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org status update
On Mar 15, 2007, at 12:32 AM, shadowym wrote: Hard to expect the business community to take Asterisk seriously when this sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives could just suddenly fail simultaneously. There must be more too it. It is drifting off topic, but if all the drives in the array where bought from the same batch, and it was a bad batch, they could all fail at about the same time. I've seen it happen in non-raid drives, I had a batch of drives all bought at the same time, that all went bad within about a week of each other. Each was in a different PC so they had slightly different up times and usage. I could see if those drives had been in a RAID array and were being stressed equally, they may have all failed within hours or even minutes of each other. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)
On Mar 7, 2007, at 9:58 AM, Steve Totaro wrote: Might as well since it is free after rebate. Just as a heads up, that rebate, like most of the others for Vonage based items, requires Vonage activation in order to actually get the rebate. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] seeing DTMF passed to Voicemail
I'm having a strange issue. My voicemail is working fine, however, any time I try to access it via one of my analog phones that are connecting to Asterisk via a Mediatrix 1124... the voicemail system complains I've entered the wrong password. There is about a 15 second pause between when I finish dialing in the password, and it complains it is wrong. This ONLY happens with phones connected via the Mediatrix. My IP phones, and soft phones all work fine and have no problems accessing voicemail. And I know the VM accounts related to the Mediatrix based phones are ok, as I can access them via an IP phone dialing into general VM access (and then specifying the box and password from there). I'm guessing that the Mediatrix is failing to send the DTMF tones correctly, or possibly send them at all. I have it set to use RFC-2833, same as my IP phones. Is there somewhere or someway to see in Asterisk either via a debug command, or in some log somewhere, what the VoiceMail system thinks is being entered? Or, has anyone else run into something similar and knows why it keeps rejecting passwords sent via the Mediatrix? -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best FXO Gateway
On Feb 15, 2007, at 1:12 AM, jameson asterisk wrote: Can anyone provide a recommendation based on user experience? Feel free to suggest an alternative gateway if one stands out. I've been working with the Grandstream GXW-4108 (the 8 port version of the 4108), and it was a rough start, but I *think* all my issues have been worked out. Initial setup wasn't too bad, it helped that I found someone else's notes on it on the Trixbox forum. It it hadn't been for two problems, I'd have probably been up and running with it in just a few hours. The two issues I had were 1: I had major logging problems that I originally blamed on the GXW, but turned out to by my syslog server. When I changed to a different syslog server, the GXW's logs started working fantastically. I needed that logging to debug the 2nd issue. The 2nd problem was more involved, the GXW didn't like my PSTN connection. It worked wonderfully when connected to my VoIP ATA, but when I went to PSTN, it had all sorts of problems. I had some back and forth dialog with Grandstream and they think they found the problem and fixed it, and sent me a beta test firmware to try out. I put that online yesterday, and so far it has been working fine, but yesterday my office was closed due to snow, so I wasn't able to really stress test it (but I also was not able to reproduce my PSTN connection problem, which previously I could do with ease, so it gives me hope that the problem is indeed fixed). I'll know better today when the office is open and I expect the 8 lines to be in use for a good part of the day with a solid mix of inbound and outbound calling. If all goes well, then I might be able to recommend the GXW-410x as a viable unit. However, it does have one feature that might be a show stopper for some. It selects the next outbound FXO port in a round robin manner. There does not appear to be port level control over which FXO port is used on a given outbound call. This is probably fine for most of the targeted users (going on price, I'd say they are aiming at the SoHo market, which is likely to have a single bank of numbers in a single hunt group, so round robin would work fine). But for some, this could be a show stopper and prevent them from being able to use the unit. I personally have to see if it is going to work for me as I actually have two different hunt groups in my 8 lines, so round robin is less than ideal for me as it can cause one group to busy out from outbound calls, while the other has no calls at all. I do plan to send a feature request to Grandstream to give better control over selecting outbound ports. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding 4 more POTS lines
On Jan 26, 2007, at 10:39 AM, Drew Gibson wrote: You can get the option numbers and values from the source html of the web page. (I am assuming the GXW-4108 works the same as other Grandstream products) I'll try that out, thanks! I did see a thread on another forum mentioning the HTML source for other products would yield the info needed, but without some kind of an example, I didn't know how exactly to turn that into what was needed. I'd assume, since the 4108 web setup looks and acts exactly like my Grandstream GXP-2000 phone, that the same trick would work for all their products. I'm certainly going to give it a try. Someone from Grandstream reached out to me last night after seeing my review post about the 4108, so when I give him a call in a little while, I'll double check that their config tool will in fact work with it as well. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding 4 more POTS lines
On Jan 25, 2007, at 5:38 PM, Leif Neland wrote: A Grandstream GXW-4108 with 8FXO cost $400, almost the same as a TDM404B fully populated 4FXO card. I'm currently testing a GXW-4108... my verdict is still out. I've had some problems, some minor, some major. In the minor department, it does not always reboot when instructed to via the web interface. I think I've tracked it to the reboot button on a regular screen is ignored, but the reboot from the post update screen goes thru. This is likely a minor bug in the firmware. Next into the minor-ish... the documentation isn't great. It it written assuming you know a lot more about this stuff than I did when I started. The more I've played and learned, when I go back and reread parts of the docs, they then make more sense to me. Heading into the not so minor, but not really major... the logging sucks. It only supports a syslog server, which isn't a huge deal, but having a web interface to read the logs would have been nice. But, the logging doesn't seem to give much info (even in Debug mode), and seems to randomly stop working entirely. Sometimes it will start again when you power cycle the unit (not just a software reboot, but physically turn it off and back on), other times it needs to be defaulted to factory settings to get the logging going again, which is totally unacceptable. Also in the not so minor category, there doesn't appear to be any easy way of backing up the config files. When it polls the tftp server on boot, it does look for a config file, but since there doesn't appear to be any way to save one out of the unit, and no documentation or otherwise (that I've found) to create one from scratch... it makes it very difficult to save settings and then easily restore them. And then into the potentially major catagory... I've run into a problem that I *think* I've tracked to the unit doesn't recognize the dial-tone issued by my PSTN provider (Verizon). It works inbound and outbound just fine at my house, where it is connected to a LinkSys PAP that interfaces with Verizon's VoiceWing service. But when I move it to a real POTS line, it works inbound, but outbound single stage dialing stalls. This is a problem that I only just identified last night, and have been working on it today and as I said I *think* it may be that it isn't accepting the dial-tone. There is an option to ignore the dial-tone and not wait, but I haven't tested that yet (at 3am I gave up at the office I was connecting it to and brought it back to my house where it promptly started working again... I'm hoping to retest on POTS tonight or tomorrow). All of the above are probably fixable via a firmware update. I'm currently running the latest that Grandstream has on their web site, but I have not yet contacted them to see if they have a newer beta version available that hasn't been publicly posted. My guess is, all the issues will be worked out in due time. With the only show stopper appearing to be the dial-tone issue (or whatever is causing it to fail on the POTS lines), it may be a good buy if you can either verify that it works with your PSTN provider first, or have the ability to return it if it doesn't (in my case, I could probably return it to the dealer, but A: it has been over a month since I bought it, and B: I'm not done playing with it to see what might or might not be wrong, and since for me price is the single most important factor, I'm willing to keep at this one to see if I can get it all working correctly.) -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to transfer calls when analog phone has no transfer button
On Jan 5, 2007, at 12:02 PM, Erick Perez wrote: When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? You can setup a dial rule to do transfering based on keypresses. I can't give you specifics on how as I'm still using Trixbox which handles some of that stuff for you. But in my setup, I was able to turn on ## as a lead in for a transfer (ie: ##205 to transfer the call to extension 205). This worked fine for softphones. When I got my Mediatrix box working correctly and connected actual analog phones, I planned to do the same, but the Mediatrix seems to override something and instead reads a hook flash as the transfer key, so on my actual analog phones, I just flash and then dial the extension and it transfers. The ## is still active for my softphones, so I think my flash setup on the analogs is something being done by the Mediatrix box itself (I think it may be generating whatever a transfer button on an IP phone does) -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. I just went thru this recently. I ended up buying a compatible modem on Ebay. You can find them easily if you search for FXO or X100 but then you may also end up paying a premium to get one that is specifically being sold to the Asterisk community. (keep in mind premium being around $30, so we still aren't talking about an outrageous price) What I did was checked the voip-info.org wiki on modem based FXOs and then searched ebay for modems listed with the correct chipsets. I lucked out and found one for $2.00 (with shipping I think it cost me $8.00 total). Mine is shows up as a Motorola X100 (or something to that effect). Seems to work fine, although I wasn't able to get Caller-ID working correctly (but I think that was a settings issue and I stopped pursuing it as it wasn't important for my pitching Asterisk). I too did this using Trixbox. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXW-4108 8 port FXO
Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using similar devices (that is, FXO devices that connect to the Asterisk server via SIP over Ethernet). I am looking to connect at least 8 PSTN lines, and as many as 12 or 16 to Asterisk (Currently using Trixbox, but I'm also looking at either AsterixNow or just building from scratch on a bare linux box). Money is a major concern in my purchases, which is why I'm looking at the Grandstream (even used on ebay, I don't seem to be able to find 8-16 port FXO devices for less than the approx $50 per port the Grandstream will get me... plus it has a video input for a security camera which is just a plus to me as installing a web capable surveillance camera at the location is on my to do list). -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Searchable Archives of this list
Is there a searchable archive of this list? Did I overlook something obvious? I can find the archives, but short of downloading all the monthly gzips and building my own searchable database, it seems my only other option is to go month by month looking at subjects and hope to stumble on what I'm after. Does anyone maintain a public searchable version of the archives? I've got tons of questions brewing, but I can't believe I'd be the first to ask any of them, so I'd really rather search thru old posts for answers before asking something that has likely been asked a dozen times before. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searchable Archives of this list
On Dec 13, 2006, at 7:42 PM, Hadley Rich wrote: Google does :) http://www.google.com/search?q=something+site:lists.digium.com Sweet... I live off of google, and for some reason trying a site specific search from google just didn't cross my mind. Thanks! -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mediatrix 1124 setup
On Dec 11, 2006, at 8:58 PM, Tim Panton wrote: It looks like there might be enough info on these pages to get you going: Thanks for the links! Hopefully I can get somewhere with the info. If you need a hand with the SNMP side, drop me a mail I'm pretty new to SNMP, so I may take you up on that once I have some intelligent questions to ask. I'll play around with it for a while and see what I can learn first. Thanks again! -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mediatrix 1124 setup
I recently purchased a Mediatrix 1124 from an auction of a company that went out of business. It came with nothing other than the unit itself. In digging thru the Mediatrix web site, and various google searches, it looks like it only supports SNMP setup, and only with their software (or the correct MIB). However, Mediatrix doesn't appear to let you download said software or MIB from their web site. Does anyone know where I can get the setup software or MIB needed to program this thing? I *think* I need the correct one for its firmware version, but I can't find out how to tell what version firmware it has. There is what appears to be the remains of a sticker marked Rev 4 on the bottom if that is any help. I have been able to default the unit and it properly gets a DHCP lease, but doesn't appear to respond to anything other than pings, a port scan reveals no open TCP ports on it. That is as far as I have gone with it so far. I figured I'd ask around for the setup software before I struggled too much more with it. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Verizon VoiceWing support
Has anyone been able to get Asterisk to work with Verizon's VoiceWing service? I'm in the process of testing Asterisk to see if it will fit the needs of my company. Since I already have Verizon's VoiceWing VoIP service, I figured if I can tie into it, that would let me evaluate service going to a VoIP provider. I've done a bunch of searching, but didn't turn up anything about how to get Asterisk to talk to VoiceWing. Verizon does not seem to officially offer anything except use of their supplied ATA (a LinkSys PAP2 that is locked down just like Vonage does... and none of the Vonage hacks seem to work on the VoiceWing one, so I can't get in and see how it is configured). I know Verizon Business offers VoIP services, including IP Trunking with the expectation that you will supply your own interface hardware. So I figured VoiceWing may be going off the same or similar systems and thus be able to support Asterisk if only the connection info was known. So, has anyone already figured this out and can point me in the right direction? -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any possibility of Vonage Integration
On Dec 6, 2006, at 8:13 AM, Paul wrote: Also, I should have mentioned that many of these providers advertise business plans on their website. How can anyone honestly advertise phone, fax, email hosting, web hosting, etc. to the business community without 24/7 support? People should also keep in mind what is the definition of 24/7 support. Even the big telcos only offer a limited amount of 24/7 support. I have dealt with business accounts with MCI (ne: WorldCom, ne: LDDS) for eons, as well as Verizon (ne: Bell Atlantic) and GTE (before and after becoming part of Verizon), and SBC (ne: PacBell). In all of the above cases, yes, they have a 24/7 support line, but when it comes to actually fixing and addressing problems, unless it was a simple issue, it waited until the next business day to actually be worked on (save for physical line problems, then Saturday was an available day as well). But any after 5pm support (or often 3:30-4:00 on Friday's for Verizon), you could call and get your trouble ticket opened, but don't expect anything to be resolved until some time after 8am on the next day. In fact, I often could get residential non business issues addressed after hours faster and easier then business issues. Once the business office closed for the day, support for business problems all but stopped until they opened again. So going with anyone over someone else because of 24/7 support, you need to find out what kind of support you really get after hours. If you are just going to get someone that can take your call, tell you why yes, that is a problem, here is your ticket number, they will work on it in the morning, then are you really gaining anything over going with someone that only offers 9-5 support? Personally, my criteria for picking someone for business use isn't if they offer 24/7 support, but rather how reliable are they in the first place. I don't care that much about 24/7 support, because I want someone who I will never have to find out what support hours they offer :-) -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
