[asterisk-users] Best way to get 60+ analogue extensions.

2009-03-15 Thread Duncan Turnbull
Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with

Re: [asterisk-users] automatic call bridging when destination is available feature

2009-03-15 Thread Olivier
2009/3/14 Vieri rentor...@yahoo.com --- On Sat, 3/14/09, Olivier oza-4...@myamail.com wrote: If I understand correctly, you're suggesting to implement the h priority instructions (or a hangup macro) to: 1) run a deadagi or a system() script to see if someone has left a request

Re: [asterisk-users] Best way to get 60+ analogue extensions.

2009-03-15 Thread Rob Hillis
Duncan Turnbull wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally

Re: [asterisk-users] Best way to get 60+ analogue extensions.

2009-03-15 Thread Stephen Davies
Hi, Xorcom make what you are looking for. Steve On 3/15/09, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions

[asterisk-users] X-Asterisk-HangupCause - how to disable this?

2009-03-15 Thread Chris Maciejewski
Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris ___ -- Bandwidth and

Re: [asterisk-users] X-Asterisk-HangupCause - how to disable this?

2009-03-15 Thread Olivier
2009/3/15 Chris Maciejewski ch...@wima.co.uk Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( For curiosity's sake, what are the

[asterisk-users] Using PRI_CAUSE to change SIP INVITE rejection response code

2009-03-15 Thread George Pajari
I am trying to change the SIP response to an incoming call and according to the docs and a quick scan of the chan_sip.c code one is supposed to be able to set the PRI_CAUSE variable and invoke Hangup but regardless of the value in PRI_CAUSE the SIP rejection is always 603. What is wrong here?

Re: [asterisk-users] Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ?

2009-03-15 Thread Olivier
2009/3/11 Tzafrir Cohen tzafrir.co...@xorcom.com On Wed, Mar 11, 2009 at 02:56:58PM -0500, Kevin P. Fleming wrote: Vieri wrote: - use the latest release of misdn v1 - upgrade to the latest stable kernel and use the built-in misdn v2 There is no support for mISDN v2 in Asterisk to

[asterisk-users] Dahdi Error

2009-03-15 Thread Julian Lyndon-Smith
Got this in the log, with no calls active. Is it a problem with my isdn line, or * ? [Mar 15 11:36:18] ERROR[29161]: chan_dahdi.c:8735 dahdi_pri_error: ACK received for '0' outside of window of '39' to '40', restarting [Mar 15 11:36:18] == Primary D-Channel on span 1 down [Mar 15 11:36:18]

Re: [asterisk-users] Using PRI_CAUSE to change SIP INVITE rejection response code

2009-03-15 Thread Steve Howes
On 15 Mar 2009, at 11:30, George Pajari wrote: I am trying to change the SIP response to an incoming call and according to the docs and a quick scan of the chan_sip.c code one is supposed to be able to set the PRI_CAUSE variable and invoke Hangup but regardless of the value in PRI_CAUSE the

[asterisk-users] Too many notify events causing Asterisk crash?

2009-03-15 Thread James Lamanna
Hi, We've implemented a 'page-all' function for some of our customers, and we've noticed that on occasion the page-all will cause asterisk to crash (safe_asterisk then restarts it again). The particular customer has about 20 phones, and also has 5 Linksys 932 to monitor the state of these

Re: [asterisk-users] Best way to get 60+ analogue extensions.

2009-03-15 Thread Jeff LaCoursiere
On Sun, 15 Mar 2009, Duncan Turnbull wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server

Re: [asterisk-users] X-Asterisk-HangupCause - how to disable this?

2009-03-15 Thread Chris Maciejewski
For curiosity's sake, what are the troubling consequences of having those headers included ? My PSTN termination provider sometimes replies with SIP/2.0 513 Message too big to my BYEs with additional headers included. Just wanted to check if this is the reason, or maybe it is related to

[asterisk-users] No hardware timing source found in /proc/dahdi

2009-03-15 Thread John Millican
Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAAGGH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version:

[asterisk-users] 428 Loop Detected

2009-03-15 Thread Asif Iqbal
Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833

Re: [asterisk-users] No hardware timing source found in /proc/dahdi

2009-03-15 Thread Shaun Ruffell
John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: /usr/sbin/dahdi_cfg If the dmesg

Re: [asterisk-users] No hardware timing source found in /proc/dahdi

2009-03-15 Thread John Millican
Shaun Ruffell wrote: John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg:

Re: [asterisk-users] No hardware timing source found in /proc/dahdi

2009-03-15 Thread Shaun Ruffell
John Millican wrote: Shaun Ruffell wrote: John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? All I see in dmesg is: dahdi: Telephony Interface Registered on major 196

Re: [asterisk-users] 428 Loop Detected

2009-03-15 Thread Steve Totaro
On Sun, Mar 15, 2009 at 6:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend

Re: [asterisk-users] Best way to get 60+ analogue extensions.

2009-03-15 Thread Duncan Turnbull
Thanks very much Rob Stephen The channel banks look good. I am not sure if they are easily availble in NZ but we can get some in I am sure. Xorom make very positive comments about their astribanks and that you can have multiple channel banks on a server so they look pretty good (if they are

Re: [asterisk-users] 428 Loop Detected

2009-03-15 Thread Asif Iqbal
On Sun, Mar 15, 2009 at 8:28 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: On Sun, Mar 15, 2009 at 6:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case

Re: [asterisk-users] 428 Loop Detected

2009-03-15 Thread Paul Hales
I am probably missing something, being a newbie. I have a 4 port fxs/fxo (2/2) card. My land line is going to one of the FXO port and my home phone is connected to one of the FXS port. I want to be able to call my phone number from external phone (cell phone) and have my home phone ring.

Re: [asterisk-users] 428 Loop Detected

2009-03-15 Thread Marco Mouta
Hi, problem is that you are saying that phone in sip.conf is at the same ip address of your asterisk box so you are dialing into a loop to your self asterisk box [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 what you need is: [phone]

Re: [asterisk-users] 428 Loop Detected

2009-03-15 Thread Jose P. Espinal
Hello Asif, I have experienced 'loop detected' when the peer where I want to send the calls to, and the asterisk Box have both the same IP address (That would make a loop). Could you please verify? Regards, Asif Iqbal wrote: Hi I looked at few emails related to this subject. And still

[asterisk-users] Asterisk 1.6 ReceiveFAX problem

2009-03-15 Thread MaxGao
hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded

Re: [asterisk-users] Asterisk 1.6 ReceiveFAX problem

2009-03-15 Thread MaxGao
asterisk 1.4.23.2 and spandsp 0.0.4 get the same error nowbut less times than other version ... [Mar 16 10:12:50] DEBUG[23749]: chan_dahdi.c:7115 do_monitor: Monitor doohicky got event Alarm on channel 1 [Mar 16 10:12:50] DEBUG[23752]: chan_dahdi.c:4731 __dahdi_exception: Exception on 11,

Re: [asterisk-users] Asterisk 1.6 ReceiveFAX problem

2009-03-15 Thread David Backeberg
On Sun, Mar 15, 2009 at 9:57 PM, MaxGao ss...@126.com wrote: and many times when reciving tax , the E1 card will down , all the channel get red alarm... [Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event Alarm on channel 2 [Mar 16 09:49:19] WARNING[20928] chan_dahdi.c:

Re: [asterisk-users] No hardware timing source found in /proc/dahdi

2009-03-15 Thread John Millican
Shaun Ruffell wrote: John Millican wrote: Shaun Ruffell wrote: John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? All I see in dmesg is: dahdi: Telephony Interface

Re: [asterisk-users] 428 Loop Detected

2009-03-15 Thread Steve Totaro
Again, if I am interpreting this correctly, he is not using SIP. A four port card 2fxo/2fxs means to me that he is not using SIP at all. If by card, you mean some kind of SIP gateway, then I misunderstood and the problem, but seeing DAHDI channels leads me to believe that SIP is not required and