Re: [Asterisk-Users] Asterisk LDAP Authentication Problem

did you notice the two dots in the IP address of ldaphost ? Marco. Chandan Mishra wrote: Hi I want to authenticate the asterisk users from the LDAP directory server not from the sip.conf. I tried to use the astirectory-1.2 http://www..asterisk-ev.org/download/astirectory-1.2-0.3.tgz . But

[Asterisk-Users] CallProgress breaks DTMF

Hi, I enabled Callprogress in the zapata.conf , so in the CDR it will log other things other then answered (Busy, no answer etc), but, this seems to break my Polycom's DTMF, i configured RFC2833 for the dtmf in the sip.conf, and when callprogress is enabled, the dtmf doesnt reach the other

[Asterisk-Users] CallProgress breaks DTMF - RFC2833

Hi I enabled Callprogress in the zapata.conf , so in the CDR it will log other things other then answered (Busy, no answer etc), but, this seems to break my Polycom's DTMF, i configured RFC2833 for the dtmf in the sip.conf, and when callprogress is enabled, the dtmf doesnt reach the other end,

[Asterisk-Users] Hangup detection - TDM400P

Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in

[Asterisk-Users] CallerID Length

Hi, I have a problem with the Caller ID string, seems like asterisk will display only 10 digits of the caller id. If the string is longer then 10 digits, asterisk will sometimes strip the first digit, and sometimes the last digits, in order to show a 10-digit callerid, Is this

Re: [Asterisk-Users] Hangup detection - TDM400P

Yes, didnt change anything Marco. Angelito Manansala wrote: hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's

[Asterisk-Users] PRI pass-through

Hi, I want to build a PRI pass-through with a Cisco 2600, with two VWIC E1 cards, is this possible ? and do i need any other modules except for the E1 modules ? What i want to do is connect the asterisk to the PRI through the Cisco router, and let my legacy PBX utilize some of the PRI

[Asterisk-Users] Detect registered peers

Hi, Is there a way to detect (in the dialplan) if a SIP peer is registered with the server ? I am using macros to dial to extension, becuase i dont want to define each extension in the dialplan, and, for example, my numbers are 8xx , i want to know if a peer exists/registered before

Re: [Asterisk-Users] fxotune fails with valid TDM/FXO card

Hi, I am using Asterisk 1.0.9 with the 1.2.0 zaptel, just for the fxotune utility, which solved my echo problems , my zttest results are low, but no echo on ZAP lines... Marco. Chris Miller wrote: Mojo with Horan Company, LLC wrote: The recent suggestion on the list was to not use

[Asterisk-Users] App_directory + Festival

Hi, As anyone tried integrating App_Directory with any Text2Speech mechanism like festival ? Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Broadvoice Outages?

Yes, i am having timeouts on registering to the LAX sip server of broadvoice. Marco. Nate Kapi wrote: I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week? ___

[Asterisk-Users] zttest - 100% ?

Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT is disabled, PCI latency was changed, i still cant get more then 99.975% in

Re: [Asterisk-Users] zttest - 100% ?

2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Marco Supino wrote: Hi, I would like to know what type of configuration could get me closer to 100% hits in zttest, when testing a TDM400P with 4 FXO ports, I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh CPU, HT

[Asterisk-Users] IBM x306 - some progress

Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows bus-centric view, as seen by

[Asterisk-Users] IBM x306

Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios

Re: [Asterisk-Users] IBM x306

Of Marco Supino Sent: Saturday, September 24, 2005 8:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IBM x306 Hi, This is a little off-topic,but if someone has any info, it could help me a LOT!, I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem

Re: [Asterisk-Users] IBM x306

Hi, I tried setpci INTERRUPT_LEVEL (or something similar, cant remmeber now), and also setpci seems like it changed the IRQ, lspci -v still shows the old IRQ Marco. Stefan de Konink wrote: On Sun, 25 Sep 2005, Marco Supino wrote: I am building an asterisk pbx (1.0.9) on an IBM x306

[Asterisk-Users] VoiceXML

Hi, Anyone has a working example of VoiceXML with asterisk ? i was looking around voip-info and the internet, and couldnt find more then proof of concept documents. Also, does anyone knows how FWD does their VoiceXML (411) service ? Thanks for any info Marco.

[Asterisk-Users] Chan_modem_*

Hi, I was looking for solutions for simple FXO cards, and came across the two modem channels in the asterisk channels/ dir, i assume they are there becuase someone made these two types of modems work as FXO (or are they there for other purpose ?), does anyone have any info on these channels ?

[Asterisk-Users] Need info : lspci

Hi, I need some info from people with the x100p card (digium or clone), please send me the output of lspci and lspci -n from your linux machine, i am tring to find out something on my * server. Thanks. Marco. ___ Asterisk-Users mailing list

Re: [Asterisk-Users] possible bug in chan_capi concerning context handling

Do you have an 's' extention in the default context ? Marco. Dimitris Kounalakis wrote: Hello, I am trying to configure asterisk 1.0.7pre to get incoming calls from an ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is that the context is not recognised in the

[Asterisk-Users] SIP registration problem

Hi, I am adding phones to my asterisk setup, until now i worked with some softphones, with no problem, I got some Grandstream BT100 phones, and see something strange in the log, the on the phone's screen, This is from the log : Found peer '122' Looking for 122 in default Transmitting (no NAT):

[Asterisk-Users] IAXTel problems

Hi, I tried to add the IAXTel config to my asterisk, so i can dial free numbers inside the US from my SIP softphone (X-lite), everything seems to be working, but the sound quality is terrible, the other side sounds like a digitized voice, and the voice is cut, i cant hear a full word, I tried