Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

Hello What will be your exact kernel version. Give me output uname -a command. -- Regards, Chandrakant Solanki On Thu, Jul 15, 2010 at 6:58 AM, Thermal Wetland thermalwetl...@gmail.comwrote: On Wed, Jul 14, 2010 at 4:55 AM, bruce bruce bruceb...@gmail.com wrote: I am stuck with the same

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

On Wed, Jul 14, 2010 at 8:09 PM, Chandrakant Solanki solanki.chandrak...@gmail.com 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

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

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 On Thu, Jul 15, 2010 at 12:00 PM, Thermal Wetland thermalwetl...@gmail.comwrote:

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki solanki.chandrak...@gmail.com 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

Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

On Wed, 14 Jul 2010, bruce bruce wrote: Thanks for the input but that won't be good because people are not going to remember two extensions for one person. People don't have to - that's what computers are for... This wouldn't be hard to do in the dialplan, but it would need some custom

Re: [asterisk-users] sip message to ip 330 or 550 phones

On Wed, 14 Jul 2010, Jerry Geis wrote: Is it possible to send a test message to the IP 330 or 550 polycom phones with asterisk? Why don't you just try it? It's one line in a dial-plan... Or have you lost the power of experimentation... Gordon --

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki solanki.chandrak...@gmail.com 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,

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

On Wed, Jul 14, 2010 at 8:43 PM, Chandrakant Solanki solanki.chandrak...@gmail.com 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,

Re: [asterisk-users] power outage

On Wed, 2010-07-14 at 23:52 -0400, C F wrote: On Wed, Jul 14, 2010 at 5:03 AM, liuxin nyliuxin...@gmail.com wrote: Hi, probably a misconfiguration or you havent plugged the cable in yet. OMG you are right, I forgot to plug in the cable. Hey but wait which cable you talking about?

[asterisk-users] How to deal with voice SMS - Asterisk 1.4

Hi list, I face a problem with voice SMSs. In some countries, if you send an SMS to a landline number, the mobile operator will record the message and then call this number. When picking up the phone you hear You get an SMS from phone number, press 1 to listen the message, 2 to repeat the

Re: [asterisk-users] How to deal with voice SMS - Asterisk 1.4

On Thu, 15 Jul 2010, Administrator TOOTAI wrote: Hi list, I face a problem with voice SMSs. In some countries, if you send an SMS to a landline number, the mobile operator will record the message and then call this number. When picking up the phone you hear You get an SMS from phone number,

[asterisk-users] Invalid host name

Hi Group, Is there anyway to force asterisk to use the ip address instead of the hostname in the sip via header. Our client's gateway is using a not FQDN as the hostname of their gateway. And I am suspecting that the asterisk is dropping the call because it could not resolve the hostname.

Re: [asterisk-users] Invalid host name

leonimar cape wrote: Hi Group, Is there anyway to force asterisk to use the ip address instead of the hostname in the sip via header. Our client's gateway is using a not FQDN as the hostname of their gateway. And I am suspecting that the asterisk is dropping the call because it

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

On Wed, Jul 14, 2010 at 03:28:35PM -1000, Thermal Wetland wrote: 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:

Re: [asterisk-users] Invalid host name

Hi Gareth, Thank you very much for the quick reply. No, I haven't tried that one since all of the gateways are on public ip address so I didn't consider enabling the NAT. But if thats how asterisk behaves when nat is set to yes then it could work. I will try it and let you know what happens.

[asterisk-users] MeetMe incorrectly reading key presses

Hi We have a few conference numbers and all use MeetMe using the D option. We have noticed sometimes that the server is picking up more key presses than were actually done, i.e. the user presses 1234 for the pin and in the logs we see something like Created MeetMe conference 1022 for

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

Try also installing the devel version of your kernel. I manage to find the link below: http://download.openvz.org/kernel/branches/rhel5-2.6.18/028stab064.7/ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm Try installing it and then recompile the dahdi module. Hope this helps. -- Mac

Re: [asterisk-users] How to deal with voice SMS - Asterisk 1.4

Le 15/07/2010 10:38, Gordon Henderson a écrit : On Thu, 15 Jul 2010, Administrator TOOTAI wrote: Hi list, I face a problem with voice SMSs. In some countries, if you send an SMS to a landline number, the mobile operator will record the message and then call this number. When picking up

Re: [asterisk-users] sip message to ip 330 or 550 phones

n Wed, 14 Jul 2010, Jerry Geis wrote: / Is it possible to send a test message to the IP 330 or 550 polycom // phones with asterisk? / Why don't you just try it? It's one line in a dial-plan... Or have you lost the power of experimentation... Gordon Gordon, I just did try it - and

[asterisk-users] Asterisk core dumping on SendFax with FFA

Hello! This has already been fixed in recent releases of FFA; there was a bug previously where the module would cause Asterisk to crash if a document to be sent could not be queued (for one of many reasons). OK, happy to hear, but when or where the recent release of FFA will be available?

Re: [asterisk-users] Asterisk core dumping on SendFax with FFA

On 07/15/2010 07:15 AM, Ilmars Knipšis wrote: Hello! This has already been fixed in recent releases of FFA; there was a bug previously where the module would cause Asterisk to crash if a document to be sent could not be queued (for one of many reasons). OK, happy to hear, but when or

Re: [asterisk-users] SKYPE - Authenticate incoming call automatically

On 07/14/2010 11:29 PM, Neeraj Chand wrote: Hi All, After getting licences for Skype for asterisk a while ago I finally got around to setting up a server with two channels and setting up a bcp on the skype end. The idea behind this is the following: Users can dial into the PBX,

[asterisk-users] centos 5 rpm pacakges (add asterisk16-xmpp module)

Hello. Who can add asterisk16-xmpp module to packages.asterisk.org or build asterisk with support xmpp and update packages? Thank You. -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- Bandwidth and Colocation

Re: [asterisk-users] sip message to ip 330 or 550 phones

On Thu, 15 Jul 2010, Jerry Geis wrote: n Wed, 14 Jul 2010, Jerry Geis wrote: / Is it possible to send a test message to the IP 330 or 550 polycom // phones with asterisk? / Why don't you just try it? It's one line in a dial-plan... Or have you lost the power of experimentation...

[asterisk-users] One way audio when dialing multiple registrations

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/x...@192.168.0.20:5060

Re: [asterisk-users] realtime music on hold

Hello, has anybody an idea or experience with this realtime moh ? Jonas. On 07/14/2010 08:53 PM, Jonas Kellens wrote: Hello list, using asterisk 1.4.30. When setting up the MySQL table 'musiconhold' as described in http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf ,

Re: [asterisk-users] One way audio when dialing multiple registrations

One-way audio is mostly firewall problem. Are you behind firewall ? You can check the audio-ports that are being used in the SDP-message by doing a /sip debug/. Maybe you do not have enough UDP-ports open for the audio ? Jonas. On 07/15/2010 04:38 PM, Nasir Javaid wrote: Hi, I am

[asterisk-users] Last call for AstriCon talks

AstriCon in Washington DC is only 102 days away! October 26-28 - slightly over three months - time is flying. The early bird discount ($595 for the whole conference) runs out next week - see if you can get in under the wire! The final selection of AstriCon talks is under way. If you've

Re: [asterisk-users] One way audio when dialing multiple registrations

Hi! I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. You need to make sure that these two phones use

Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

Yes, thanks. I think lots of manufacturers just boost the number of speakers really needed but again this really depends on the environment noise level. Regards, Bruce On Wed, Jul 14, 2010 at 11:50 PM, C F shma...@gmail.com wrote: I'm happy to hear it worked out so well with so little. :) On

Re: [asterisk-users] centos 5 rpm pacakges (add asterisk16-xmpp module)

On 07/15/2010 08:16 AM, Vasiliy G Tolstov wrote: Hello. Who can add asterisk16-xmpp module to packages.asterisk.org or build asterisk with support xmpp and update packages? Thank You. This is something we've been considering for a while. It should make its way onto the list shortly. --

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

El 15/07/10 04:34, Tzafrir Cohen escribió: On Wed, Jul 14, 2010 at 03:28:35PM -1000, Thermal Wetland wrote: 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:

[asterisk-users] Good script to make appointment?

Hello I'd like to write a script that would make it easier for people to call in, listen to the IVR, and make an appointment (eg. When? ASAP? A given day? - Morning? Afternon, etc.) I assume I'm not the first one to try and write this type of IVR, so would appreciate any feedback on writing

[asterisk-users] WARNING[15867]: chan_sip.c:15766

Hello Asterisk-Community, I'm having an error with my BLF configuration on my asterisk...i've configured the sip peer like this: [8250] type=friend callerid=Extensión 8250 8250 canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup=

Re: [asterisk-users] BLF with Realtime

Thanks as always Zeeshan ;) I've changed my configuration, take a look: [8250] type=friend callerid=Extensión 8250 8250 canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250

[asterisk-users] QoS and Asterisk

I have discussed QoS with our ISP and in order to implement this, I need to make sure all VoIP packets are marked in the IP packet header (IPP bits?). Does Asterisk automatically marks the VoIP packets or do I need to do something in Asterisk? I need to do this for SIP and H323 protocols.

[asterisk-users] beeping during call

On Wed, Jul 14, 2010 at 09:27:29AM -0700, Steve Casto wrote: Asterisk 1.4.32 dahdi-2.3.0.1 Centos 5.5 Digium TE420 CAC channel bank (2) Cisco RVS4000 router Cox 50 Mbps/ 5 Mbps cable modem Flowroute provider codac G-711 90 % CPU idle callwaiting=no When there are 10-15 or

Re: [asterisk-users] beeping during call

On Thu, Jul 15, 2010 at 10:19:10AM -0700, Steve Casto wrote: https://issues.asterisk.org/view.php?id=17529 Thanks Tzafrir: Unclear on how to apply patch, here is what I get: [r...@localhost asterisk-1.4.32]# patch -p1 ../bug17529.diff.txt can't find file to patch at input line 5 Perhaps

Re: [asterisk-users] Good script to make appointment?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, July 15, 2010 11:40 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Good script to make appointment? Hello I'd like to

[asterisk-users] Asterisk Manager Problem

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/d...@outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID:

Re: [asterisk-users] Asterisk Manager Problem

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deric Page Sent: Thursday, July 15, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Manager Problem I am originating a call to a

[asterisk-users] Does Flash Operator Panel allow for dragging a call into a parking lot?

Hi Everyone, If I receive a call on a ZAP line and pickup the call and drag and drop it (by mouse) into a Parking Lot through FOP, it just hangs up. Is this feature supported by FOP? Thanks, Bruce -- _ -- Bandwidth and

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

On Wed, Jul 14, 2010 at 11:34 PM, Tzafrir Cohen tzafrir.co...@xorcom.com 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

[asterisk-users] Soft-phone on Black Berry

Hi All, i have a question, is there any soft-phone available for Black Berry use, I've been told there is a firefly one, but when i looked, i found nothing, is any body has an update on this please? -- _ -- Bandwidth and

Re: [asterisk-users] QoS and Asterisk

On 07/15/2010 11:13 AM, hin lee wrote: I have discussed QoS with our ISP and in order to implement this, I need to make sure all VoIP packets are marked in the IP packet header (IPP bits?). Does Asterisk automatically marks the VoIP packets or do I need to do something in Asterisk? I need

[asterisk-users] help with sip registration

I am sending SIP registration and I get 200 OK messages back however something is not right . In /var/log/asterisk/messages I am getting : [Jul 15 16:40:42] NOTICE[2875] chan_sip.c:-- Registration for '5...@10.164.112.3' timed out, trying again (Attempt #33) Its not correctly registering. -

[asterisk-users] Dahdi T1 CRC4 errors?

I have a system setup with two T1 circuits, the customer complains of call quality issues. I've checked logs and don't see anything too strange, but when I run dahdi show status I see: Description Alarms IRQbpviol CRC4 Fra Codi Options LBO T2XXP (PCI) Card

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

El 15/07/10 15:15, Thermal Wetland escribió: On Wed, Jul 14, 2010 at 11:34 PM, Tzafrir Cohen tzafrir.co...@xorcom.com 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

Re: [asterisk-users] Soft-phone on Black Berry

Someone whom I know at blackberry's software development team, says blackberry doesn't support VoIP at this time. So I doubt if there is any working VoIP client for it. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-15 4:19 PM, khalid touati khalidtou...@gmail.com wrote: Hi All, i have a

Re: [asterisk-users] Soft-phone on Black Berry

-- * GI (NOT?) YF * Try this link * http://voip.about.com/od/mobilevoip/a/BlackBerryVoIP.htm -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Soft-phone on Black Berry

Blackberry has a very high dollar proprietary solution for what you are trying to achieve, I don't think they ever allow SIP soft-phones on their devices. -- Jeremy Betts (714) 388 6015 Ext. 304 Freevoice -- _ -- Bandwidth

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

On Thu, Jul 15, 2010 at 11:29 AM, Miguel Molina mmol...@millenium.com.co wrote: El 15/07/10 15:15, Thermal Wetland escribió: On Wed, Jul 14, 2010 at 11:34 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: The DAHDI Makefile looks for the kernel source by default in:  /lib/modules/`uname

Re: [asterisk-users] Good script to make appointment?

On Thu, 15 Jul 2010 12:39:51 -0500, Danny Nicholas da...@debsinc.com wrote: This how I would do it Thanks a lot Danny. I'll study this and see how it goes. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] SKYPE - Authenticate incoming call

Hi All, After getting licences for Skype for asterisk a while ago I finally got around to setting up a server with two channels and setting up a bcp on the skype end. The idea behind this is the following: Users can dial into the PBX, get authenticated and only after

Re: [asterisk-users] SKYPE - Authenticate incoming call

On Friday 16 Jul 2010, Neeraj Chand wrote: Hi All, After getting licences for Skype for asterisk a while ago I finally got around to setting up a server with two channels and setting up a bcp on the skype end. The idea behind this is the following: Users can dial into the PBX,

[asterisk-users] IAX endpoints not Registering after upgrage from Asterisk ver 1.4.26.1

Dear All, I am experiance a issue with my IAX clients. I have upgradeed Asterisk to 1.4.28 After then IAX clients are not working and It's not registering even. Please help. Asterisk previous version - 1.4.26.1 ( for this worked fine) FreePBX version - freepbx-2.5.2 -- Thanks Regards,

[asterisk-users] Queue

hi, all Is ther any way to set 3-way conference using queue app other other way using queue app. scenario: custmore call to queue , agent answered than agent transfer to third persion, so the three call communicate with each other. Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD)