[asterisk-users] Unknown connection error: (2006) MySQL server has gone away

2007-08-29 Thread Shaun Hofer
Hi, I get the following after a call has finished: ERROR[6862]:mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away Does this error message only appear when asterisk makes a new connection to mysql, because the old connection was stale (and dropped) ? If so, is

Re: [asterisk-users] TDM400 hungup problem

2006-11-05 Thread Shaun Hofer
Ale, We had a simiarly problem here, not sure if its the same. The Telco here has 'ISDN suspension' (think thats the correct term) activated on landlines here by default. When you phone some one, the person who recieves the call can put down the reciever and goto another room and pick it up

[asterisk-users] column width in CLI

2006-08-23 Thread Shaun Hofer
Hi, Can the column width for commands run in the Asterisk CLI be increased? Currently when I run 'show channels' I can't see the whole channels id/name as its to long for the columns width and is cut off. I need to grab a list of active channels, which is currently not do able. Thanks Shaun

[asterisk-users] sip host and registering

2006-08-15 Thread Shaun Hofer
I have a setup where in sip.conf the host=ser.zxy.com for the phones. Non of the phones are connected to Asterisk directly, but are connected to SER. Thus non of the phones registry with Asterisk. I have noticed that when I forward a call to Asterisk it doesn't send the call back to SER (which

Re: [asterisk-users] SIP trunks: order or type

2006-08-14 Thread Shaun Hofer
you 2006/8/11, Rich Adamson [EMAIL PROTECTED]: Shaun Hofer wrote: ok maybe I can explain my problem better. There two trunks both have the same details except one is type=peer (and only does ulaw) and the other is type friend (and does ulaw/alaw/g729). Incoming calls should

Re: [asterisk-users] SIP trunks: order or type

2006-08-14 Thread Shaun Hofer
On Monday 14 August 2006 21:18, Rich Adamson wrote: Shaun Hofer wrote: It wasn't any help. It doesn't give any reference to order of trunks, etc in sip.conf. I'm still looking for the post, Rich Adamson made reference too... On Friday 11 August 2006 18:24, Fran Oliveira wrote: see

Re: [asterisk-users] SIP trunks: order or type

2006-08-10 Thread Shaun Hofer
, this trunk must be type=peer or type=friend To inbound calls to * box via SIP trunk , this trunk must be type=user or type=friend. friend=user+peer peers cannot place calls into the Asterisk server http://www.asteriskpbx.com/ Best regards, Marco Mouta On 8/10/06, Shaun Hofer [EMAIL

[asterisk-users] SIP trunks: order or type

2006-08-09 Thread Shaun Hofer
I have two trunks to the same machine (x.x.x.2), one is type=friend, other is type=peer. Asterisk seems to choose which trunk to use by the order by which they are set out in sip.conf. When a incoming call comes into Asterisk, it always uses the last trunk. My understanding was that a peer

[asterisk-users] softhangup() problem

2006-08-01 Thread Shaun Hofer
I have been trying to test out softhangup(). Every time I use it in a macro, it doesn't seem to hang up any call/s on the trunk. I have used: exten = s,1,SoftHangup(SIP/trunk-sx) exten = s,1,SoftHangup(SIP/trunk-sx|a) exten = s,1,SoftHangup(SIP/trunk-sx-1) exten =

Re: [Asterisk-Users] OPENSER / SER and Asterisk

2006-06-13 Thread Shaun Hofer
www.onsip.org One of the best places for ser info -Shaun On Wednesday 14 June 2006 11:44, Kelvin Williams wrote: Has anyone ever published a concise howto or good documentation on how the two interrelate? and Configurations.. On 6/13/06, BILL GITONGA [EMAIL PROTECTED] wrote: Asterisk

Re: [Asterisk-Users] GXP-2000

2006-06-06 Thread Shaun Hofer
I suggest you contact grandstream about this. Only thing I can suggest is look at feature's Early Dial (I have set to no) and No Key Entry Timeout (set to 10-15 seconds). As for all these other problems of phone stop working, etc., we haven't come across these in office (then again we don't

[Asterisk-Users] gxp-2000 Asterisk PSTN

2006-05-08 Thread Shaun Hofer
Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it

[Asterisk-Users] Grandstream GXP-2000 call end

2006-05-02 Thread J Shaun Hofer
Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use

[Asterisk-Users] codec variable for incoming calls

2006-04-24 Thread J Shaun Hofer
Hi, In Asterisk, is there way to find out which codec is being used by incoming call? Is there some variable or function call that can be done? Thanks -- Shaun *** If you receive this email by mistake, please notify us and do not make any use

[Asterisk-Users] sip.conf codecs: ulaw, alaw and g729

2006-04-19 Thread J Shaun Hofer
Hi, When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw) stop working and I get the frame type error for them, but g729 works fine. I've cleared general part of sip.conf of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Asterisk