[asterisk-users] Polycom getting DCHP address from wrong VLAN
Hello, I have been tearing my hair out on this issue for 2 days, any help would be appreciated. We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch There are two VLANs, 1(data) & 50(VoIP). When Polycoms are connected to the switch with VLAN 50 hard coded in the config they grab a DHCP address from VLAN 1, the PVID for the switch port. The ports have membership in VLAN 1 as the PVID and VLAN 50 as tagged traffic. I know the VoIP DHCP server is working because if I change a port to have a PVID of 50 any device gets the address from the VoIP DHCP server. I have tried the ports as 'general' and 'trunk' with no success. Any help would be greatly appreciated, I don't have much hair left! -- -Thermal -- _ -- 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] Can't compile DAHDI - wrong kernel source
On Thu, Jul 15, 2010 at 11:29 AM, Miguel Molina wrote: > El 15/07/10 15:15, Thermal Wetland escribió: > > On Wed, Jul 14, 2010 at 11:34 PM, Tzafrir Cohen > wrote: > > > The DAHDI Makefile looks for the kernel source by default in: > > /lib/modules/`uname -r`/build > > This is normally a symlink that points to the directory with the > (possibly partial) kernel source tree. > > It seems that the kernel package you have installed did not set up this > symlink. You can create it yourself. > > Alternatively, set set KSRC explicitly to the (full path to) the kernel > tree: > > make KSRC=/usr/src/ > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > > Creating the link was the answer! Even when I installed the RPM the > link in /lib/modules was not created. > > After make all, make install, and make config DAHDI will not start. > in the /drivers/dahdi folder I only have dahdi_dummy.c > All the other files have .ko .mod.c .o etc > Is this related? > > [r...@ip-97-74-119-59 dahdi-linux-complete-2.3.0.1+2.3.0]# dahdi_cfg -v > DAHDI Tools Version - 2.3.0 > Notice: Configuration file is /etc/dahdi/system.conf > line 0: Unable to open master device '/dev/dahdi/ctl' > > 1 error(s) detected > > The only warning I see is "WARNING: could not find > /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd > for > /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o" > > Does that ring a bell with any one? > > > > I experienced the same issue, and googling about this, on some openvz > documentation they are clear: for security you can't enable kernel modules > in an openVZ VPS (you can't even do a lsmod, it shows nothing), the only way > I could use DAHDI timing in my case for asterisk inside a VPS, was to > install DAHDI in the HN (Hardware Node) and share it using this config > settings: > > vzctl set 101 --devnodes 'dahdi/pseudo:rw dahdi/channel:rw dahdi/timer:rw > dahdi/ctl:rw' --save > > Where 101 is your CTID (container ID). > > Look at http://forum.openvz.org/index.php?t=msg&goto=37706&; > > That way I made it! Now I have a single DAHDI instance in the HN shared into > two VPS, and dahdi_cfg and dahdi_test works great: > > [r...@virtual1_ast1 ~]# dahdi_test > Opened pseudo dahdi interface, measuring accuracy... > 99.951% 99.641% 99.572% 99.604% 99.671% 99.972% 99.979% 99.991% > --- Results after 8 passes --- > Best: 99.991 -- Worst: 99.572 -- Average: 99.797645, Difference: 99.998742 > > The DAHDI timing is used without issue by asterisk: > > virtual1_ast1*CLI> module show like timing > Module Description Use > Count > res_timing_dahdi.so DAHDI Timing Interface > 484 > res_timing_pthread.so pthread Timing Interface 0 > > virtual1_ast1*CLI> timing test > Attempting to test a timer with 50 ticks per second. > Using the 'DAHDI' timing module for this test. > It has been 1016 milliseconds, and we got 51 timer ticks > > > My goal was to provide DAHDI timing to asterisk inside a VPS. I don't know > if having a TDM card in the HN configured with DAHDI it can be shared so a > VPS can access it and then you could receive PRI calls or something directly > into a VPS, or better, split channel groups between VPS. > > Cheers, > > -- > Ing. Miguel Molina > Grupo de Tecnología > Millenium Phone Center Thank you Miguel for the openVZ help! -- -Thermal -- _ -- 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] Can't compile DAHDI - wrong kernel source
On Wed, Jul 14, 2010 at 11:34 PM, Tzafrir Cohen wrote: > The DAHDI Makefile looks for the kernel source by default in: > > /lib/modules/`uname -r`/build > > This is normally a symlink that points to the directory with the > (possibly partial) kernel source tree. > > It seems that the kernel package you have installed did not set up this > symlink. You can create it yourself. > > Alternatively, set set KSRC explicitly to the (full path to) the kernel > tree: > > make KSRC=/usr/src/ > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir Creating the link was the answer! Even when I installed the RPM the link in /lib/modules was not created. After make all, make install, and make config DAHDI will not start. in the /drivers/dahdi folder I only have dahdi_dummy.c All the other files have .ko .mod.c .o etc Is this related? [r...@ip-97-74-119-59 dahdi-linux-complete-2.3.0.1+2.3.0]# dahdi_cfg -v DAHDI Tools Version - 2.3.0 Notice: Configuration file is /etc/dahdi/system.conf line 0: Unable to open master device '/dev/dahdi/ctl' 1 error(s) detected The only warning I see is "WARNING: could not find /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd for /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o" Does that ring a bell with any one? -- -Thermal -- _ -- 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] Can't compile DAHDI - wrong kernel source
On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki wrote: > Hi > > Following steps to do... > > 1] # cd /usr/src/kernels/ > 2] # ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7 > > Try this 'n let me know... Hope this will work fine... > > > -- > Regards, > > Chandrakant Solanki > I have all the extra kernels because I ran 'yum install kernel*' Could that be the issue? -- _ -- 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] Can't compile DAHDI - wrong kernel source
On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki wrote: > Hi > > Following steps to do... > > 1] # cd /usr/src/kernels/ > 2] # ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7 > > Try this 'n let me know... Hope this will work fine... > > > -- > Regards, > > Chandrakant Solanki > Chandrakant, Are you curious enough to want shell access to take look? I am going to blow this away and start again once I learn the secret! I have done so much experimenting. -Thermal -- _ -- 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] Can't compile DAHDI - wrong kernel source
On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki wrote: > Hi > > Following steps to do... > > 1] # cd /usr/src/kernels/ > 2] # ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7 > > Try this 'n let me know... Hope this will work fine... Seems like that should have worked! [r...@ip-97-74-119-59 kernels]# ls -l total 117492 drwxr-xr-x 21 root root 4096 Jul 14 18:40 2.6.18-128.2.1.el5.028stab064.7-i686 drwxr-xr-x 19 root root 4096 Jul 14 19:35 2.6.18-164.11.1.el5-PAE-i686 drwxr-xr-x 19 root root 4096 Jul 14 19:36 2.6.18-164.11.1.el5-debug-i686 drwxr-xr-x 19 root root 4096 Jul 13 20:25 2.6.18-164.11.1.el5-i686 drwxr-xr-x 19 root root 4096 Jul 14 19:36 2.6.18-164.11.1.el5-xen-i686 lrwxrwxrwx 1 root root 28 Jul 14 19:35 2.6.18-164.11.1.el5PAE-i686 -> 2.6.18-164.11.1.el5-PAE-i686 lrwxrwxrwx 1 root root 30 Jul 14 19:36 2.6.18-164.11.1.el5debug-i686 -> 2.6.18-164.11.1.el5-debug-i686 lrwxrwxrwx 1 root root 28 Jul 14 19:36 2.6.18-164.11.1.el5xen-i686 -> 2.6.18-164.11.1.el5-xen-i686 drwxrwxr-x 20 root root 4096 Jul 14 18:45 linux-2.6.18.8 -rw-r--r-- 1 root root 52488302 Jul 14 18:16 linux-2.6.18.8.tar.gz drwxrwxr-x 24 root root 4096 Jul 14 19:05 linux-2.6.34.1 -rw-r--r-- 1 root root 67658955 Jul 5 11:27 linux-2.6.34.1.tar.bz2 [r...@ip-97-74-119-59 kernels]# ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7 [r...@ip-97-74-119-59 kernels]# ls -l total 117492 lrwxrwxrwx 1 root root 36 Jul 15 00:03 2.6.18-028stab064.7 -> 2.6.18-128.2.1.el5.028stab064.7-i686 drwxr-xr-x 21 root root 4096 Jul 14 18:40 2.6.18-128.2.1.el5.028stab064.7-i686 drwxr-xr-x 19 root root 4096 Jul 14 19:35 2.6.18-164.11.1.el5-PAE-i686 drwxr-xr-x 19 root root 4096 Jul 14 19:36 2.6.18-164.11.1.el5-debug-i686 drwxr-xr-x 19 root root 4096 Jul 13 20:25 2.6.18-164.11.1.el5-i686 drwxr-xr-x 19 root root 4096 Jul 14 19:36 2.6.18-164.11.1.el5-xen-i686 lrwxrwxrwx 1 root root 28 Jul 14 19:35 2.6.18-164.11.1.el5PAE-i686 -> 2.6.18-164.11.1.el5-PAE-i686 lrwxrwxrwx 1 root root 30 Jul 14 19:36 2.6.18-164.11.1.el5debug-i686 -> 2.6.18-164.11.1.el5-debug-i686 lrwxrwxrwx 1 root root 28 Jul 14 19:36 2.6.18-164.11.1.el5xen-i686 -> 2.6.18-164.11.1.el5-xen-i686 drwxrwxr-x 20 root root 4096 Jul 14 18:45 linux-2.6.18.8 -rw-r--r-- 1 root root 52488302 Jul 14 18:16 linux-2.6.18.8.tar.gz drwxrwxr-x 24 root root 4096 Jul 14 19:05 linux-2.6.34.1 -rw-r--r-- 1 root root 67658955 Jul 5 11:27 linux-2.6.34.1.tar.bz2 [r...@ip-97-74-119-59 kernels]# cd /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/ [r...@ip-97-74-119-59 dahdi-linux-complete-2.3.0.1+2.3.0]# make all make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 [r...@ip-97-74-119-59 dahdi-linux-complete-2.3.0.1+2.3.0]# -- _ -- 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] Can't compile DAHDI - wrong kernel source
On Wed, Jul 14, 2010 at 8:09 PM, Chandrakant Solanki wrote: > Hello > > What will be your exact kernel version. Give me output "uname -a" command. > > -- > Regards, > > Chandrakant Solanki > Thank you for the help! Here is the output: [r...@ip-97-74-119-59 ~]# uname -a Linux ip-97-74-119-59.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux -Thermal -- _ -- 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] Can't compile DAHDI - wrong kernel source
On Wed, Jul 14, 2010 at 4:55 AM, bruce bruce wrote: > > I am stuck with the same problem but I have used asterisk yum repository and > it worked by itself without me worrying for kernel stuff. > However, I need to install speex codec and now I am stuck as it doesn't get > picked up by the yum asterisk install somehow. I have lib speex and speex > already installed and when doing "yum install asterisk16" I don't see speex > in "core show translation" Is there anything specific I have to do? > Do I have to build from source as well? > -Sorry, didn't mean to hijack the thread. > Thanks, > Bruce > On Wed, Jul 14, 2010 at 5:08 AM, Chandrakant Solanki > wrote: >> >> Hi >> >> If you install rpm from any location it goes to its default location. >> >> You just go for above steps. For kernel you can go for http://kernel.org >> >> -- >> Regards, >> >> Chandrakant Solanki >> >> On Wed, Jul 14, 2010 at 2:06 PM, liuxin wrote: >>> >>> Hi. >>> The best easy way is: >>> copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src >>> then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm >>> >>> 2010/7/14 Gareth Blades >>>> >>>> Thermal Wetland wrote: >>>> > I have a virtual server with godaddy but can not compile DAHDI as it >>>> > complains that I do not have the correct kernel source. >>>> > >>>> > The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: >>>> > Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and >>>> > latest version >>>> > Nothing to do >>>> > >>>> > uname -a returns: >>>> > Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net >>>> > <http://ip-XXX-XXX-XXX-XXX.ip.secureserver.net> 2.6.18-028stab064.7 #1 >>>> > SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux >>>> > >>>> > When I try to compile DAHDI it fails with: >>>> > make[2]: Leaving directory >>>> > `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' >>>> > You do not appear to have the sources for the 2.6.18-028stab064.7 kernel >>>> > installed. >>>> > >>>> > Is there a way to trick DAHDI to use the installed kernel? >>>> > >>>> > Thanks for the help! >>>> > >>>> > -- >>>> > -Thermal >>>> > >>>> >>>> What kernel versions do you have installed? >>>> >>>> If you are currently running an older kernel but installed a newer >>>> kernel and sources but havent rebooted to activate the new one yet then >>>> it may still be trying to locate the source for the older running kernel. >>>> >>>> >>>> I was able to download the rpm's and install them: [r...@ip-97-74-119-59 src]# rpm -ivh ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm warning: ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm: Header V3 DSA signature: NOKEY, key ID a7a1d4b6 Preparing... ### [100%] package ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686 is already installed [r...@ip-97-74-119-59 src]# rpm -ivh ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm warning: ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm: Header V3 DSA signature: NOKEY, key ID a7a1d4b6 Preparing... ### [100%] package ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686 is already installed [r...@ip-97-74-119-59 src]# cd - /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0 [r...@ip-97-74-119-59 dahdi-linux-complete-2.3.0.1+2.3.0]# make all make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 The directories in /usr/src/kernels is: [r...@ip-97-74-119-59 kernels]# ls -l total 51328 drwxr-xr-x 20 root root 4096 Jul 14 18:04 2.6.18-128.2.1.el5.028stab064.7-i686 drwxr-xr-x 19 root root 4096 Jul 13 20:25 2.6.18-164.11.1.el5-i686 drwxrwxr-x 19 root root 4096 Feb 23 2007 linux-2.6.18.8 I tried to install the kernel from source but couldn't find the exact kernel, I installed linux-2.6.18.8 as I was the closest. Both of the directories in /usr/src/kernels/ have the -i686 suffix, is that the issue? -- -Thermal -- _ -- 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] Can't compile DAHDI - wrong kernel source
I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux When I try to compile DAHDI it fails with: make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. Is there a way to trick DAHDI to use the installed kernel? Thanks for the help! -- -Thermal -- _ -- 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] Put a call on hold with Manager
I would like to be able to place a call on hold via the manager interface and be able to retrieve it. The user can click a button in the Order entry form to put the caller on hold when they are looking up information. It saves them from having their hands leave the keyboard and press hold on the phone. I don't see 'hold' & 'retrieve' commands for the manager interface. -- -Thermal -- _ -- 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] Interesting One Way Audio
On Tue, Apr 13, 2010 at 7:43 PM, Prince Singh wrote: > > >1. Are Asterisk and Mittel in the same physical LAN.. or do they have a >router between them? >2. Do a 'rtp debug' at the Asterisk CLI to see where is the RTP data >being sent to >3. Probable issues:- > 1. canreinvite is enabled when it should not be > 2. Mitel is sending SDP with an incorrect RTP IP and/or port... > You'll need to check 'sip debug' to see what RTP port is being sent >4. From the 1/2 second audio, it seems that it could be due to one of >these:- > 1. 1/2 second is early media, and is being handled correctly at both > Mitel and Asterisk. OR, > 2. After 1/2 second, Asterisk and Mitel renogotiate for RTP payload > type, and switch to a codec that is broken at either or both the > locations > 3. After 1/2 second, Asterisk and Mitel renogotiate for RTP IP/port > > > In case you are unable to debug with the above help, post these:- > >1. IPs of both Mitel and Asterisk >2. SIP dialog as text (sip debug output should do) >3. A few lines of RTP debug output > > -- > Regards, > Prince Singh > > Drishti-Soft Solutions Pvt Ltd > > Thank you for the feedback, 4.1 about early media led me to the answer! Your ideas and voip-info.org searching helped! my extensions.conf was like this: answer() cut() dial() I changed it to: cut() dial() Thanks again for your assistance! -Thermal -- _ -- 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] Interesting One Way Audio
I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user dials a number that autoanswers line 1-800-555-1212 the Mitel user will hear audio for 1/2 a second then it is dropped. I turned of iptables and it acts the same way. Anyone have any ideas? -- -Thermal -- _ -- 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] Force Jitter Buffer for SIP to SIP calls
On Wed, Dec 30, 2009 at 8:27 AM, Matt Darnell wrote: > > # Asterisk sip jbenable = yes|no : Enables the use of a jitterbuffer > on the receiving side of a SIP channel. (Added in Version 1.4) > # Asterisk sip jbforce = yes|no : Forces the use of a jitterbuffer on > the receive side of a SIP channel. Defaults to "no". (Added in Version > 1.4) > > It mentions the 'receiving side' which should be the incoming or > upload form the clients. > As I am sure you saw, it is not mentioned in the peers and clients section. > Perhaps setting jbforce to no and jbimpl to adaptive. > > I am sure you read all that, anyone have any real world experience? > > Aloha, > Matt > Thank you for confirming that I was reading it correctly. I will be looking at the SPA-2102 to see if it can do anything in regards to how it is transmitting voice. -- -Thermal ___ -- 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] Force Jitter Buffer for SIP to SIP calls
We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want to enable the jitter buffer for the end points having the trouble. Reading the docs, it seems that the jitter buffer is only used when the end point is connected to an app like voicemail. -- -Thermal ___ -- 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] Context Switches and Load Average spike - Asterisk Version 1.4.22
I am running Asterisk V 1.4.22 Twice during the last two days the Context Switches on our box has gone from about 7K to 80K in 2.5 hours. The load average would spike to 17, drop to 0.35 then spike again. When connecting to the console 'core show channels' will list the channels but not total calls. 'restart now' had no effect, the only way to stop Asterisk is to kill the process. Once Asterisk is killed, everything returned to normal, for about 20 hours, then it started again. The server is a dual - quad core machine. Linux has been up over 380 days. Has anyone experienced this before? -- -Thermal ___ -- 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] Valet Park with Hint - Button Support
We have Valet Park working well with 1.4.25. We have programmed the Polycom softkeys to include a park button that does a blind transfer to the park extension. Has anyone gotten the a button to activate when a particular park orbit is in use? It would be great if you could press the button to retrieve the parked call. -- -Thermal ___ -- 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] Is anyone keeping up with the versions?
We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt ___ -- 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] Polycom's lose BLF after Asterisk restart
We have an issue where Polycom's lose BLF functionality after a reboot. The only way to fix it is to reboot the Polycoms. Anyone else have this issue? We are using 1.4.18. If I run 'sip show subscriptions' all the subscriptions come back after the restart but the lights on the phones do not work. Any help would be appreciated. -Thermal ___ -- 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] Looking for a Snom expert
On Thu, May 8, 2008 at 5:20 AM, Brent Davidson <[EMAIL PROTECTED]> wrote: > Which phones are you using and what software revision. I've had a crash > course in Snom phone lately and can probably help with at least the park > orbits. > > -Brent > Brent, We have the phones in the lab, we have 1 - 370, 1 - 360 & 2 - 320. If we can make them work, we will certainly purchase more. I find their software/manuals very confusing. I can't find any user guide for the 370, the CD that came with the phone had a manual for the 360. We would be able to use any software version that worked the best/stable. I will be willing to compensate someone for their time. Thermal ___ -- 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] Looking for a Snom expert
On Wed, May 7, 2008 at 7:45 PM, Paul Hales <[EMAIL PROTECTED]> wrote: > > Where are you located? > We are located on the west coast. The person could work remotely, and we would pay (I should have said that in the first email!) Thermal ___ -- 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] Looking for a Snom expert
I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal ___ -- 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] Is there a distro with hlyafax rolled in?
On Fri, Apr 4, 2008 at 4:42 PM, Jonn R Taylor <[EMAIL PROTECTED]> wrote: > I made some install scripts based on centos 4 or 5 like trixbox but > without all the junk. It does have some fax setup stuff in it that I use on > our production servers that's been working for over a year. I you need any > help you can email me directly. > > > > Jonn > > > > http://www.taylortelephone.com/asterisk/ > > Jonn, The install script looks great. I will try it on Monday. Thanks! ___ -- 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] Is there a distro with hlyafax rolled in?
On Fri, Apr 4, 2008 at 12:55 PM, James Finstrom < [EMAIL PROTECTED]> wrote: > elastix > thank you james for the recommendation, I will check it out. ___ -- 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] Is there a distro with hlyafax rolled in?
Or any fax solution? On Fri, Apr 4, 2008 at 11:46 AM, Thermal Wetland <[EMAIL PROTECTED]> wrote: > Does anyone know of a Trixbox like install that has the hylafax > integration rolled in? > > Looking for basic fax to email support. > > Thanks, > Thermal > ___ -- 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] Is there a distro with hlyafax rolled in?
Does anyone know of a Trixbox like install that has the hylafax integration rolled in? Looking for basic fax to email support. Thanks, Thermal ___ -- 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] Anyone know of a pass through ATA
That is good to know. I will be using the the device to connect a bunch of analog phones back to a centralized server. Since each office has only one ethernet connection do I want to: 1. Install a small switch to let a SPA1001 & PC use the one drop 2. Install a SPA3102 to let the PC share the one drop Even with an extra patch cord and probably the occasional power strip number one will be cheaper, but #2 seems like a better way to go. -Thermal On Fri, Mar 14, 2008 at 6:41 PM, <[EMAIL PROTECTED]> wrote: > its not a bad device - I have 2 problems with it. It doesn't do echo > cancellation very well & is particularly badly matched to the PSTN here in > Oz. Hint: keep it well cooled - echo goes up badly when its hot& it runs > very hot if there is no ventilation. I use the 3102 to bridge a mythtv box > instead of putting in an extra switch - works except for the occaisional > failure to get a dhcp address. I use a linux gateway for dhcp, most devices > (3102+mythtv box, lynksys PAP2, bt100 ip phones, wireless and hosts) are all > dhcp > > -Original Message- > > From: "Thermal Wetland" <[EMAIL PROTECTED]> > Subj: Re: [asterisk-users] Anyone know of a pass through ATA > Date: Sat 15 Mar 2008 10:23 > Size: 2K > To: [EMAIL PROTECTED], "Asterisk Users Mailing List - Non-Commercial > Discussion" > > That is awesome. I dont know why the manual doesnt mention that. > > I want to have the device use a static IP & the computer use DCHP from a > central DHCP server...sounds like it wont be a problem. > > Thanks. > > On Thu, Mar 13, 2008 at 8:14 PM, W.Kenworthy <[EMAIL PROTECTED]> wrote: > sipura 3102 set to bridge. Works but I find that when rebooting a PC > bridged it sometimes (randomly) doesnt get a dhcp lease, necessitating a > powercycle of the 3102. I think the PC drops the ethernet as it reb > oots and the sipura doesnt recognise it coming back. > > BillK > > > > On Thu, 2008-03-13 at 19:59 -1000, Thermal Wetland wrote: > > Anyone know of a company that makes a pass through ATA? > > > > By pass through I mean have an Ethernet switch built into the ATA, > > like most desktop phones have. > > > > All of the dual ethernet ATAs I have seen have WAN/LAN ports, not two > > LAN ports. > > > > I fooled around with DMZ etc...but it just doesnt work as well. > > > > Thermal > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone know of a pass through ATA
That is awesome. I don't know why the manual doesn't mention that. I want to have the device use a static IP & the computer use DCHP from a central DHCP server...sounds like it won't be a problem. Thanks. On Thu, Mar 13, 2008 at 8:14 PM, W.Kenworthy <[EMAIL PROTECTED]> wrote: > sipura 3102 set to bridge. Works but I find that when rebooting a PC > bridged it sometimes (randomly) doesnt get a dhcp lease, necessitating a > powercycle of the 3102. I think the PC drops the ethernet as it reb > oots and the sipura doesnt recognise it coming back. > > BillK > > > On Thu, 2008-03-13 at 19:59 -1000, Thermal Wetland wrote: > > Anyone know of a company that makes a pass through ATA? > > > > By pass through I mean have an Ethernet switch built into the ATA, > > like most desktop phones have. > > > > All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two > > LAN ports. > > > > I fooled around with DMZ etc...but it just doesn't work as well. > > > > Thermal > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Group Listen on SIP Phone
Anyone know of a SIP phone that supports group listen? Group listen allow you use the handset but what the far end says comes out the speaker...it is F802 on a Norstar. Thermal ___ -- 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] Anyone know of a pass through ATA
Anyone know of a company that makes a pass through ATA? By pass through I mean have an Ethernet switch built into the ATA, like most desktop phones have. All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN ports. I fooled around with DMZ etc...but it just doesn't work as well. Thermal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom - Buddy Watch not a choice when adding Speed Dial
> > > Check your sip.cfg for the line: > feature.1.name="presence" feature.1.enabled="1" > > I would imagine that you have enabled="0" > That was it! Thanks - Thermal ___ -- 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] Polycom - Buddy Watch not a choice when adding Speed Dial
Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at "Auto Divert".I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add 1 manually to the speed dial file it doesn't change anything. The buttons work well for a speed dial. The icon next the speed dial is 10 dots, in the shape of a keypad. Anyone else experience this? Thanks, Thermal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom MWI's will not turn off
On Nov 28, 2007 11:26 AM, Bruce Komito <[EMAIL PROTECTED]> wrote: > I have seen this with Polycoms, ZIP2s and occassionally with Linksys 941s, > but only intermittently. Sometimes a powercycle will clear it and > sometimes not. We've never figured out what's going on, but we think it > is something to do with NAT and the phones not exactly sticking to the > spec, but that's only a presumption. > > Bruce Komito > WPTI Telecom > (775) 236-5815 > > Bruce, Thanks for the kick in the butt! It was the NAT device...I have been chasing this for three days. It's the NAT, It's the NAT, It's the NAT! Thermal ___ --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] Polycom MWI's will not turn off
Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog ' [EMAIL PROTECTED]' in 32000 ms (Method: NOTIFY) Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5060: NOTIFY sip:[EMAIL PROTECTED]:33475 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0c3d9f34;rport From: "anonymous" ;tag=as33238a01 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 88 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) <--- SIP read from y.y.y.y:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0e58862b;rport From: "anonymous" ;tag=as69473f09 To: ;tag=D888A873-3AA22F98 CSeq: 112 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0130 Content-Length: 0 Everytime the phone re-registers these messages are sent and the phone 'beeps' and will turn the MWI indicators on even if they have been manually turned off. Anyone see the issue or have any suggestions? Thanks, Thermal ___ --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] Is the 1.0.X branch vulnerable to the SIP issue?
We are still using 1.0.7 and did not see any patches for the 1.0.X branch. Does anyone know if that branch is affected? -Thermal ___ --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] Multiple TE110P cards in one chassis
Does anyone know if you can have multiple TE110P cards in one chassis?-Thermal ___ --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] Queue reporting seems broken.
On 5/3/06, Joe Dennick <[EMAIL PROTECTED]> wrote: On the wiki, there is a little perl script that can be used to parse the queue log and insert the data into a database. I've modified the script to use a MySQL database. With that, I have a cron job that shuts down Asterisk, parses the queue-log into the MySQL database, and then restarts Asterisk (thus re-initializing a new (empty) queue log). Once the data is in the database, its pretty easy to find the information you are seeking. Simply query the database for all records (COUNT(*)) where action = '' and date is between the range you are searching for to find out how many calls entered the queue for that time-period.If you do a similar query for action = '' or 'COMPLETEAGENT' you will see all of the answered calls. You can further limit that query by specifying a particular agent. I've wrapped all of these queries in PHP and provide web forms (html) to allow the Call Center Manager to select what data to report on (date ranges, queue, agent, etc.). I can send you some of the PHP code if you are interested. In summary, once you've got the data in a database, you can extract it in any way that's meaningful to you. The wiki provides really good information about what ACTIONS are recorded, and then what information is provide in the info1, info2, and info3 fields for each ACTION. The database structure is very flat and easy to work with, so you really don't have to know very much about databases to achieve excellent reporting results. Johann <[EMAIL PROTECTED]> wrote the May 3, 2006 8:26 AM:I will check out the script to see what you are parsing for. I am sure that will tell us if fields are missing. If we can get it working, I would love to see the PHP code!Unfortunatly we can't shut down *, we have customers with 24/7 call centers.-Matt ___ --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] Queue reporting seems broken.
On 5/3/06, Johann <[EMAIL PROTECTED]> wrote: The wiki page doesn't mention the other fields in the file.:::Depending on the action some of the fields may contain 'NONE' instead. You will need to use the field to match who the connect andcompleteagent/completecaller messages are for.Keep in mind the field will be the same for a caller as they gothrough the queue. So the enterqueue, connect, complete actions will have the same. We are on version 1.07, I wonder if that is the difference.I will set up a box with the latest version and see if the output to queue_log is different.-Matt ___ --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] Queue reporting seems broken.
I am trying to figure out which one of our agents is answering the calls.According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the only time the queue_log puts the channel (agent) is during logoff & logon.There is the connect & completeagent message, but it doesn't show which channel (agent) answered the phone.I can't even figure it our cross referencing the CDR records, the CDR record only has the queue number. Is there a way around this?Aloha,Matt ___ --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
