Re: [Asterisk-Users] Clipping on outbound calls via SIP/IAX

Hi It could be that the RTP sessions aren't completely setup when you get connected to the destination. Kind Regards Claus Futtrup - Original Message - From: Jens Vagelpohl [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com

Re: [Asterisk-Users] Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help

Hi, 1) 0.0.0.0 just means listning on all interfaces and their ip adresses, not a problem. 2) Do a set verbose 100 to see if you have any communication with the sip phones or startup asterisk with asterisk -vvvggg 3) This is because a MPG3 file used for music on hold isn't support or that

[Asterisk-Users] DTMF from TE410P to SIP devices doesn't work

Hi Guys, I have this problem that when a caller from my E1's get's connected to a SIP devices and sent DTMFs. the SIP device is unable to detect the DTMF's sent. You can hear a very small portion of the DTMF sent, but not enough for the device to tell what it is. If the call is between

Re: [Asterisk-Users] eyebeam

Hi there, switch off G711 alaw codec then it should ok Kind regards Claus - Original Message - From: Altus Syman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 23, 2004 3:37 PM Subject: [Asterisk-Users] eyebeam

[Asterisk-Users] Problem with stuttering on TE410P

Hi Guys,Im having some problems with a Wildcard TE410P card.. During a call I getsome strange messages and the voice drops out:Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]:

[Asterisk-Users] problem with E100P

of span 2 No message like this for span 1.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.752 / Virus Database: 503 - Release Date: 03-09-2004 ___ Asterisk

[Asterisk-Users] Harddisk noise on TE410P

Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup

Re: [Asterisk-Users] Harddisk noise on TE410P

Claus Futtrup This message is for the designated recipient only and may contain privileged or confidential information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. - Original Message

[Asterisk-Users] Problem with sound on Wildcard TE410P

Hi Guys, Im having some problems with a Wildcard TE410P card.. During a call I get some strange messages and the voice drops out: Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Aug 24 16:40:17 DEBUG[1101416512]:

Re: [Asterisk-Users] Question about TE405P

I'll need somethine like this span=1,1,0,ccs,hdb3,crc4bchan=1-15bchan=17-31dchan=16 span=2,1,0,ccs,hdb3,crc4bchan=32-46bchan=48-62dchan=47 Setting timing source to 0 onspanscould give you some problems.. (at least I've had them) --- channel =

[Asterisk-Users] Problem with EuroISDN E1

only seem to affect span 2. Users have been complaining about being unable to make calls, but Im not sure if this has anything to do with that.. Please help. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version

Re: [Asterisk-Users] Problem with EuroISDN E1

Here you go. loadzone = no defaultzone = no span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 span=2,2,0,ccs,hdb3,crc4 bchan=32-46 bchan=48-62 dchan=47 Both E100P are connected to PSTN. Kind Regards Claus Futtrup - Original Message - From: Storer, Darren [EMAIL PROTECTED

Re: [Asterisk-Users] User-Oriented Management of Asterisk

I would be very interested in this, please let me know if I can help Claus Futtrup Project Manager The box said 'Requires Windows 95, NT, or better,' so I installed Linux. This message is for the designated recipient only and may contain privileged or confidential information. If you have

Re: [Asterisk-Users] Howto: Use setgroup, checkgroup to check incoming and outgoing client limits

Well that works.. But lets say I wont to be able to control incoming and outgoing limits on all channels. I have3 phones registered and phone 1 calls phone 2. With the example below phone 1 cannot make anymore calls.. But phone 2 can (even though stíll talking with phone 1) Phone 2 can also

[Asterisk-Users] Howto: Use setgroup, checkgroup to check incoming and outgoing client limits

Hi there, I was wondering how I can use setgroup and checkgroup for perfom incoming and outgoing limitation checks. I've have some users that doesn't what to be able to recieve more than 1 call at a time, and I also want to limit a users outgoing call abilities. Any help would be greatly

[Asterisk-Users] Problem with incominglimit and outgoinglimit

Hi, I seem to have a problem with chanisavail and the call limits on sip phones(incoming and outgoing) The problem seems to be that chanisavail when trying create to create channels and hanging them up afterwards screw up the current usage limit on the phones. Example with chanisavail: Phone

Re: [Asterisk-Users] Problem with incominglimit and outgoinglimit

von Klitzing [EMAIL PROTECTED] To: Claus Futtrup [EMAIL PROTECTED] Sent: Wednesday, June 23, 2004 4:08 PM Subject: Re: [Asterisk-Users] Problem with incominglimit and outgoinglimit Use SetGroup() and GetGroupCount() and CheckGroup() insetad of incominglimit Cheers, Philipp --- Outgoing mail

Re: [Asterisk-Users] Unable to create channel - CVS Broken?

Hi Guys, Same problem here with latest CVS. -cf Hi, Just started to get this error after updating to the latest CVS. Asterisk dies if it can't create a channel - not so good. -- Executing SetCallerID(SIP/750-2550, 39660426) in new stack -- Executing Dial(SIP/750-2550,

Re: [Asterisk-Users] Problem with SIP softphone

Hi! X-Lite: Menu -- Advanced settings -- Audio -- Silence set keep transmitting after silence to 1 or something like that Cf - Original Message - From: Philipp von Klitzing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 21, 2004 11:24 AM Subject: Re: [Asterisk-Users]