[asterisk-users] trunk peer not registering after migrating installation

2008-11-27 Thread John Taylor
I have an odd problem. I have just installed asterisk on an ubuntu box, and migrated the previous configuration of asterisk (on another ubuntu box) to this new server (scp -pr [EMAIL PROTECTED]:/etc/asterisk/* /etc/asterisk/) Asterisk worked fine on the old server, but on this server my SIP trunk

[asterisk-users] call queuing not detecting caller hang up when call originates from voip provider

2007-12-28 Thread John Taylor
Dear all I've got call queuing working when calls originate from my local site. After testing I migrated it to calls originating from our voip provider- it should ring an extension, then queue . All works well apart from if the caller hangs up when queued: the call hangs around in the queue as a

[asterisk-users] multiple sip trunks

2009-12-03 Thread John Taylor
I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than

Re: [asterisk-users] multiple sip trunks

2009-12-11 Thread John Taylor
Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want

Re: [asterisk-users] multiple sip trunks

2009-12-11 Thread John Taylor
: On Fri, Dec 11, 2009 at 10:23 AM, John Taylor j...@vetsurgeon.org.uk wrote: Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? yes, Asterisk does support multiple

Re: [asterisk-users] multiple sip trunks

2009-12-11 Thread John Taylor
I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? John 2009/12/11 Noah Miller noahisaacmil...@gmail.com: I assume if all the

Re: [asterisk-users] multiple sip trunks

2009-12-15 Thread John Taylor
,1) exten = 31592123457,1,Goto(trunk1,s,1) exten = 31592123458,1,Goto(trunk1,s,1) exten = 3159212,1,Goto(trunk2,s,1) exten = 31592123334,1,Goto(trunk2,s,1) exten = 31592123335,1,Goto(trunk2,s,1) On 14 dec 2009, at 10:39, Olle E. Johansson wrote: 11 dec 2009 kl. 23.21 skrev John

[asterisk-users] caller getting cut off intermittently

2010-01-04 Thread John Taylor
I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP --Asterisk server--Natted

Re: [asterisk-users] DNS reload on trunks for outgoing calls

2010-01-04 Thread John Taylor
Put the commonly used domain names + appropriate ips into /etc/hosts? John 2010/1/4 Steve Howes steve-li...@geekinter.net: On 4 Jan 2010, at 08:34, Remco Barendse wrote: Is there any fix or workaround for the DNS problem (old standing bug that when the box starts and domain names do not

Re: [asterisk-users] caller getting cut off intermittently

2010-01-07 Thread John Taylor
We're now getting this problem on outgoing calls. I've forced the port to 100FD but still no joy. Anyone any ideas how to debug this- have added verbose to logger.conf Thanks for any help John 2010/1/4 John Taylor j...@vetsurgeon.org.uk: I have recently moved our asterisk server from our LAN

Re: [asterisk-users] caller getting cut off intermittently

2010-01-19 Thread John Taylor
Hi, I've now set dtmfmode=rfc2833 and that seems to have fixed it John 2010/1/7 John Taylor j...@vetsurgeon.org.uk: We're now getting this problem on outgoing calls. I've forced the port to 100FD but still no joy. Anyone any ideas how to debug this- have added verbose to logger.conf

Re: [asterisk-users] caller getting cut off intermittently

2010-01-19 Thread John Taylor
Hi all, I've now set dtmfmode=rfc2833 instead of inband and that seems to have fixed it John 2010/1/4 John Taylor j...@vetsurgeon.org.uk: I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our

[asterisk-users] forward call back up same trunk to external cell phone problem

2010-01-30 Thread John Taylor
Hi If I have an incoming call coming down a SIP trunk to a particular internal SIP extension- I can answer the extension fine, all works well However, if I change extension.conf from dialling the internal extension to forward the call to an external cell phone (up the same trunk as the incoming

Re: [asterisk-users] forward call back up same trunk to external cell phone problem

2010-02-01 Thread John Taylor
Hi- can anyone help with this. I'm really stuck as apparently it should work. Is it a problem with the ITSP, with using the same trunk for both legs of the call etc? John On 30 January 2010 08:57, John Taylor j...@vetsurgeon.org.uk wrote: Hi If I have an incoming call coming down a SIP trunk

[asterisk-users] Problem attended transfer with ilbc

2010-06-28 Thread John Taylor
I have an Asterisk server on our LAN that serves our office VOIP phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are ulaw/alaw We use attended transfer extensively. If our trunk is ulaw/alaw they work fine. If the trunk is ilbc we have problems 1- incoming PSTN call routed via

[asterisk-users] Snom phones recommended firmware

2010-09-04 Thread John Taylor
We're using firmware 7.3.30 on an installation of Snom 300 phones. Should we stick with it, or do the newer firmwares have better support for Asterisk? Thanks John -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Record() Cmd and My SQL

2010-09-24 Thread John Taylor
Why not write the file to /tmp using MixMonitor, then use the command option to trigger an AGI script that will move the data into your database then delete the original file? John On 24 September 2010 04:23, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: The reason is when

[asterisk-users] tcpdump auto stats script

2010-09-24 Thread John Taylor
Before I reinvent the wheel, I'm looking for a script then when run will - launch tcpdump (or equivalent) on the server and capture all SIP and UDP traffic to an IP address - then, rather than me manually analysing with wireshark, will analyze the cap file and produce stats on jitter, lag, delta

[asterisk-users] cannot answer incoming calls

2011-01-06 Thread John Taylor
Have recently installed some Snom phones into an office. Phones are natted and connect to a 1.4 server on a public IP We can make outgoing calls, but are unable to answer incoming calls. The phone rings, but the call cannot be picked up. Other phones on other sites connected to the server are

[asterisk-users] intermittent problem on 1.4

2011-01-19 Thread John Taylor
We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline-voipfone-asterisk 1.4-voipfone-UK landline About 1 in 3 times the call at the final landline is silent

[asterisk-users] vigor 2920 problems

2011-11-21 Thread John Taylor
One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing

Re: [asterisk-users] vigor 2920 problems

2011-11-21 Thread John Taylor
Message - From: John Taylor j...@vetsurgeon.org.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 21 November, 2011 10:20:14 AM Subject: [asterisk-users] vigor 2920 problems One of our clients has a Draytek Vigor 2920- their natted

[asterisk-users] atx timeout - play xferfailsound

2012-01-30 Thread John Taylor
Asterisk 1.6.2.20 on Debian Lenny I'm finding that if no one answers an attended transfer (timeout set by atxfernoanswertimeout), then the transferrer is handed back to the original caller, and a beep is played. In 1.4 I was able to indicate the timeout and failure by setting xferfailsound to a

Re: [asterisk-users] vigor 2920 problems

2012-01-30 Thread John Taylor
Thanks for help- suggestion fixed the issue John On 21 November 2011 11:25, John Taylor j...@vetsurgeon.org.uk wrote: Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will get permission to try new firmware later! JT On 21 November 2011 10:45, Arthur Stanfield

[asterisk-users] Asterisk 1.8 not playing parking slot announcement to parker

2012-10-25 Thread John Taylor
Just upgraded to 1.8, we use the multi lot parking feature by dialling *4. We are not getting the parking slot announcement being played to the person who parks the call, so it's impossible to tell which slot they've gone into. Could someone check our config? On Debian Squeeze using packages from