similar to: Overlap digits...

Hello all, I have a problem with overlap dialing and don't know how to get rid of it. My setup is: 1 HFC card with bristuff -> ZAP/g1 (2B + 1D channels), SIP phones (I just removed TDM400P with 4 FXS) I created test extension 222 which goes directly to g1. In Zapata.conf overlapdial is set to yes. First I created this extension: exten => 222,1,Dial(zap/g1,100,tc)

Hi, I'm trying to make an apparently simple thing work, but I don't see how it is possible with Asterisk. This is my extensions.conf: exten => 1234,1,Dial(SIP/123456/555-555-5555|20|D(7777)) ;After call connects, send DTMF 7777 exten => 1234,2,VoiceMail(1234@context); What I obviously want is that if nobody answer the call, go to voicemail. Basic stuff. Problem is Asterisk

Hi, our Asterisk is connected to an E1 port. So we are using the DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for overlap digits for in-calls? I found the option "overlapdial=yes" but I did not try yet. Is that "my" option? Is there any option for setting an timeout? Thorsten

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

Hi, I've noticed that a small but meaningful quota of calls from my Alcatel PBX to Asterisk are failing. This does not always happen and it is not easily reproducible but on high traffic I do get a large number of cases. Example: Alcatel PBX extension 7085 calls Asterisk PBX extension 6145 over a PRI E1 link. I see this in the Asterisk log: Dec 14 14:10:31 VERBOSE[11378] logger.c: --

'lo, A provider sets up an Asterisk box in order to service the needs of a small number of customers. The provider issues SIP handsets and the users register with sip.telco.com Thanks to the selection of a brilliant family of technologies, including SIP and Asterisk, the telco.com company grows and grows. Eventually, beyond the point that they can really hold all of the customer SIP

P.H.B. is insisting on having the ability to create a transparant SIP tunnel between old style ISDN telephony PBX with overlap dialing: PBX - ISDN - IAD - SIP - * - DAHDI - PRI The idea is that dialed numbers a the PBX are transmitted to the PRI as they are typed, whenever the PRI gets the signal that the number is complete the dialer instantly gets a ringing. This behavior is described in RFC

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

Hi, I am new to Asterisk and am runing asterisk 1.2.9.1 on an OpenBSD box. With a few manuals I was able to set up some SIP providers with which outgoing and incoming calls work. However, there is one provider with which inbound calls don't work at all. The only apparent error/warning message is this WARNING[13688]: chan_sip.c:3527 process_sdp: Unable to lookup host in c= line, 'IN IP4

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

Steve Casto escribi?: >/ I am trying to retrieve the cause code of a outgoing call over a PRI />/ where the number called is out of service. When an out service number is />/ called I get a recording that the number dialed is not a working />/ number. I see cause code 1 in in the CLI as soon as the call is dialed />/ the Telco recording goes on for 30 sec. then hangs up. Any

Hi list, What paremeter can I change to control interdigit timing? Because my PSTN provider aren't receiving all the digits I dialed on Zap/g1. My Zap/g1 are an E1 (E400P) using E&M immediate sigalling. thanks in advance Eduardo

As an employee in the technical operations of a CLEC this information is easily obtainable by anyone that has access to the Class 5 switch servicing that PRI... A Q.931 trace in the Class 5 Switch will tell the whole story.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich Adamson Sent: Tuesday, June

Hmmm, perhaps I am the only one who doesn't trust their telco (I doubt it) but... I have the rates that I currently pay my telco, and would like to extract my CDR's and add an additional field displaying the actual price paid for the call. I would like to do this based on destination phone number, and outgoing channel. However, I have a few difficulties: 1) I pay a different rate for

Eeeeck! Asterisk is inserting random digits in dialed numbers. So far I've seen it insert a 2 after the STD (area) code and insert an extra 6 or 7 in the STD code. It's pretty repeatable although the inserted number changes. My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02). There's an ISDN PBX on the second span and a BRI euroisdn on the first. Calls from the

Hello, i have a little Problem with compiling asterisk-addons the failure is: app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: ?SR_LIST_REMOVE' undecalred (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is

I have successfully built and made asterisk talk SIP extension to SIP extension, read all the docs, and about 1000 emails from the archive. The trunk side of Asterisk, from the docs perspective, is a smidgin TDM-centric, Analogue, T1, zaptel.conf etc..... Asterisk cares not about the externally presented digits as the telco KNOWS which time-slot or analogue line the call came from I live in an

We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI & DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are

Hello List! We have problems setting the right caller id on outgoing calls. The Asterisk Pbx is located in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the local telefon number 40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID numbers available. The telco is aspecting a 3 digit long Callerid from us, for example like "710", for the extension 10.

Hi, Perhaps there's something wrong in my config... I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting up an h323 trunk. When dialling into asterisk I got some problems when the entire number is not in the setup message, i.e. I'm dialling digit by digit on the ericsson phone. Lets say I have 4001 in my extensions, and dial that from the Ericsson PBX, then the