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Skipped content of type multipart/alternative-------------- next part -------------- Console logs from Asterisk A: Executing Dial("SIP/test0-5821", "Zap/6/327557670||Tt") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 6/327557670 -- Zap/6-1 is proceeding passing it to SIP/test0-5821 -- Accepting UNAUTHENTICATED call from 195.66.73.122:

I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked

Every so often, and it seems that it happens only when a call is in progress, all 24 Zap channels get reset. All channels are opened and then timeout. This causes the in-progress calls to terminate. There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk although we do receive a fair amount of Blue Alarms. The Asterisk server is connected to a legacy PBX through a Digium

Hi, We have a Ericsson BP250 Phone system setup witht he following configuration Telco <-> Asterisk E405P <-> BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next

Hi, I've been experiencing trouble with my DAHDI channels for some time and have finally decided to try and resolve the issue. Essentially, the problem I am having is that when a call comes in, and my DAHDI phones therefore ring, Asterisk thinks that one of the handsets has picked up to answer the incoming call - whereas in actual fact it is still on hook. The call then gets instantly

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten => s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

Hi list, why I cannot hear voice when I call from a SIP telephone (Budgetone and others) to a H323 telephone (several models)? could anybody please give any idea to solve this issue? Please, let me know. Thanks in Advance. N.B. The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are: ***************************************

Hi, I guess something's wrong with my chan_dahdi configuration, ... but I can't seem to get it. When I test incoming calls on a DAHDI-channel (incoming from pstn), asterisk seems to interpret it as a caller hangup after each ring. Any ideas. OUTPUT: -- Starting simple switch on 'DAHDI/5-1' -- Executing [s at from-pstn:1] *Verbose*("*DAHDI/5-1*",

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

I have a system with two 4 port T1 cards, with 5 PRI's configured. Each PRI is configured as an individual PRI and belongs to it's own group (groups 1-5) This system is handling roll-over from another system, where any error in processing the call on that system results in it being sent here. This mainly results in all calls resulting in a busy being sent for retry here. I then

Hi, A most peculiar and vexing problem for you all. I hope I have been verbose enough without being a firehose ;) The set up: I have a channel bank, using the r1t1 rhino driver with a rhino T1 card (the channel bank itself is a very legacy piece of equipment)- this supplies FXS for all the house phones. Also, a Wildcard TDM400P, using the wctdm module with 1 FXO module, this supplies FXO to the

I can't seem to retrieve a parked call. Here is what I do: Call from phoneA to phoneB Answer phoneB Press Flash/Callwait on phoneB Press 700 to park the call A voice says that the call is parked at 701 When I try to dial 701, I don't get connected to the parked call Below is the asterisk output when I tried to park

SBC installed a T1 ADTSe (Digital Trunking Service Enhanced) e&m wink start with 24 1 way trunks. The CO says they dial pulse DP the seven digit dnis number. The channels work now but take long time to answer and get these messages repeating until I guess the CO stops Pulse dialing the number. Nov 23 19:08:58 WARNING[1827865]: chan_zap.c:4718 ss_thread: getdtmf on channel 8: Operation now

I'm trying the latest bri 0.1.0 RC2 drivers. In announce I see implementation of so long waited Transfer feature. But I can't make it work. When the person who is making transfer after talking with second party press "R" second time to establish 3 way call the person to which call supposed to be transfered being disconnected. Any ideas whats wrong? Thanks, Dmitry

Hello List, I'm using a dialstring like the one below. I want to have three different things happening depending on exit cause. Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000])) These 3 things could happen: 1, Caller hangs up 2, Callee hangs up 3, The 20 seconds is up and call is terminated from Asterisk. Is there a way to separate these 3? Thanks, Best regards, Tobias --------------

Calling out on PSTN over a TDM400P seems to generate a Red Alarm - whatever that is. I have another extension on the PSTN, and I can dial out over that. zttool shows no alarms. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup