similar to: Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

Hi List, This is somewhat off-topic since the problem itself isn?t asterisk but the Ericsson BP250 I want to connect to. But since there have been a couple of posts relating in part to that system I am hoping someone can help me out. What we want to do: PRI <---> BP250 <---> Asterisk Currently the BP250 handles extension 000 - 899. We now would like to forward a chunk of 100

hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set

hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with

hi, after several problems getting the right callerid on a E1-PRI there is (so far) only one problem left: when receiving calls over the telephone network from another E1-PRI that has a "Caller ID no screen" capability (e.g. a bank and a customer of us), asterisk does not get the callerid that is set up by the calling PBX, but the callerid of the trunk of the calling PRI. no matter

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

I noticed a problem this morning with our cdr logging. We have a cron job that places a call file into the spool directory having asterisk call itself to check to make sure its still handling incoming calls correctly, then queries the CDR database in mysql and makes sure that appropriate records exist. I can confirm that the call is happening correctly, but I'm missing records in the

Hi All I am using asterisk version cvs-v1-0-04/15/05 and spandsp 0.0.2pre18. I can receive and then email most faxes without issues, but recently I am having this issue when receiving faxes from a particular person. I can receive the faxes ok, but there are alot of bad rows as indicated by my logs below and the fax is not readable. I have included a good (from another user) and a bad fax. We

Hi All I am using asterisk version cvs-v1-0-04/15/05 and spandsp 0.0.2pre18. I can receive and then email most faxes without issues, but recently I am having this issue when receiving faxes from a particular person. I can receive the faxes ok, but there are alot of bad rows as indicated by my logs below and the fax is not readable. I have included a good (from another user) and a bad fax. We

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips.

Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the incoming calls for the 20 undefined numbers to a common resource (ivr, operator, or canned message) without having to define each one?

Hi all, We are running Asterisk 1.0.0 with a TE410P. Very often we exerience calls dropping in the middle of the call. I enable the full logging and saw a couple of suspicious messages right before the hangup. Thos could happen on a Zap-IAX2 bridge as well as on a Zap-Agent bridge... I see Nov 23 09:08:36 DEBUG[-1274020944]: Bridge stops because we're zombie or need a soft hangup:

I'm running * with a bog standard HFC ISDN card using zaphfc. Everything seems to work, including incoming calls, but I simply cannot make outgoing calls. This is very odd since the same card worked with the same configuration in another server. This is what I get from * debug. The only possible difference between the two servers that I can think of is that the HFC card is sharing an IRQ

Hello guys, I have an Asterisk installed (last version), and sometimes I have my calls hanged when I'm still talking to someone. I put the maximum debug to have informations about this problem and I found only one thing : Mar 10 01:26:16 DEBUG[16136]: Requesting Hangup because the busy tone was detected on channel Zap/1-1 Mar 10 01:26:16 DEBUG[16136]: Got a FRAME_CONTROL (5) frame on channel

Hi! We have some problems with calls that get disconnected in the middle of a call. We are using Asterisk 1.2.1 with a TE410P (2.gen firmware). When the call is disconnected Asterisk writes this to the log: Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy, avgtone: 300, avgsilence 2090 Jan 9 14:56:17 DEBUG[4404] dsp.c: Requesting Hangup because the busy tone was detected on

Hello, I'm using asterisk-1.0.8 with BRI and spandsp-0.0.2_pre20. Modules app_txfax.so and app_rxfax.so are compiled and loaded sucessfully. It seems that the channel cant't detect the call as a fax-call 7612022801 is the calling faxmaschine 1209259 is my recieving fax extension logs are:

Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everybody, I am trying to use SIP (Sipura 2000) to connect to Asterisk which then dials out a local number using the Digium E100P. We have purchased the G729 codec licenses from Digium and loaded them into Asterisk successfully. However, the call drops immediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a