similar to: CAPI ptp does not work

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |

Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi

Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10

Hello, we had a running configruation where asterisk passed the phone number and the ddi to the pstn (i.e. 595-431) Now only the rootnumber arrives: 5950 I do not know, what to do. I tried to use callingpres (now i am just hiding every number, because 595-0 is no valid extension..) but that did not worked. > Protocol Discriminator: Q.931 (8) len=44 > Call Ref: len= 2 (reference

Hi there, today I made the German language prompts available for download: http://www.karl.aegee.org/asterisk.nsf/HT/sound-de Be aware: Asterisk doesn't yet fully support languages other than English, there are still (smaller) issues with voicemail and date/time announcements that require a patch. Cheers, Philipp

Hello, I am using chan_capi 0.3.5 and Asterisk CVS-v1-0-12/29/04-15:32:48 on a SuSE Linux 9.0 with Kernel 2.4.21-99-default In the system is a AVM C4 with one port connected to PSTN at PTP BRI and another one to an ISDN PBX with an PMP BRI. The system is running fine, but I have regualary this error, and then chan_capi is not working anymore. Jan 18 15:29:46 WARNING[2919]: Avoided initial

Hi all, A new version of DIAX (0.9.8a) is ready to be downloaded from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.8a: - unconditional autoanswer or based on CallerID (user configurable); - use any Ericsson/SonyEricsson GSM/PCS to control DIAX (feedback on the phone display) through Bluetooth (or serial cable). You do not even

O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps telling me while startup: [chan_capi.so] => (Common ISDN API for Asterisk) Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr1 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr2 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr3 Jan 24 14:30:47

Hi, i use chan_capi 0.3.1 with asterisk (stable branch cvs) and 3 x c4 active ISDN card. From Controller 1 - 7 there are no problems making calls between asterisk and the pstn. But when i make calls from controller 8 - 12 i get on every controller (8 - 12) a segmentation fault in asterisk :( I tried different linux distributions (gentoo 2004.1, redhat 9.0 , suse 9.1) but same error.

Hello all, the scenario: Carrier ----S2M------ * -----S2M------Siemens | | SIP Clients and many other features With much help from the list, the PRI links are without alarms and inbound calls are working fine (from both: Carrier and Siemens). But I am not able to dial wether outbound nor to the Siemens PBX. I allways get the message: == Everyone is busy

http://www.linuxjournal.com/article/8993 If you''ve got feedback, requests for coverage, etc. please let me know. I''ll try to respond in an upcoming column. -- thanks, -pate ------------------------- http://on-ruby.blogspot.com

Hello, We have 4 'Onramp-2' Telstra ISDN BRI services operating on Asterisk Server with Eicon 4BRI card. For most part the service is okay. However, we are are having problems with passing callerID to internal extensions. This is the set of command executed. exten => <pattern>,1,Answer ; Answer the line exten => <pattern>,2,NoOp(${DNIS}) ; debug statements exten =>

Hi All, I have 2 NICS in this system. CentOS 6.4 eth0 is the virtual IP from PFSense mapping connected to the router, works fine. eth1 is a second NIC that I have assigned a private IP to and connected it to a switch on the private network. I have many other private devices, so I know this setup works. When I connect the cable to the switch and bring up eth1 the system basically stops taking

Hello All! I have a strange problem with Cisco 7905. It is forwarding unanswered calls to VoiceMail even thought I have setup it not to. My ring timer on cisco 7905 is 60s, and my ForwardToVMDelay is 3000s. This means that call should never be forwarded to VM! This is true if I call from internal number then this happens on asterisk: -- SIP/104-6073 is ringing -- Nobody picked up in

From: Vivek Thampi <vithampi at vmware.com> Date: Fri, 28 Feb 2020 05:32:46 +0000 > Add a PTP clock driver called ptp_vmw, for guests running on VMware ESXi > hypervisor. The driver attaches to a VMware virtual device called > "precision clock" that provides a mechanism for querying host system time. > Similar to existing virtual PTP clock drivers (e.g. ptp_kvm),

From: Vivek Thampi <vithampi at vmware.com> Date: Fri, 28 Feb 2020 05:32:46 +0000 > Add a PTP clock driver called ptp_vmw, for guests running on VMware ESXi > hypervisor. The driver attaches to a VMware virtual device called > "precision clock" that provides a mechanism for querying host system time. > Similar to existing virtual PTP clock drivers (e.g. ptp_kvm),

Good day, does anyone have AstTAPI running for incoming calls, and would like to show some examples. My setting right now looks like this: sip.conf -------- [22] type=friend dtmfmode=info username=22 mailbox=22 secret=privat host=dynamic context=privat canreinvite=yes callgroup=1 incominglimit=2 extension.conf -------------- exten => 123,1,noop ;Hint(SIP/22) exten =>

WARNING[<pid>]: CAPI[contr3/123456]/178 already has PBX structure?? WARNING[<pid>]: CAPI[contr3/123456]/178 already has a call record?? WARNING[<pid>]: CDR on channel 'CAPI[contr3/12345]/177' already started WARNING[<pid>]: Thread 1109916592 Blocking 'CAPI[contr3/123456]/178', already blocked by thread 1116277680 in procedure ast_waitfor_nandfds

hello, i change my isdn line to ptp mode and i have some trouble to configure chan_capi in that way. What is the correct setup - e.g. if numbers are 123456x - where x can be from 0 to 9 ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hi ! I've fetched a spare AVM B1 card from the cellar, and installed it. After "modprobe b1pci" I did "capiinit" and capiinit moaned about a missing t1.b4. So I search the web and found one at http://www.avm.de/ftp/cardware/b1/x_misc/ddi/. When I now look at the controller, I finally see p2p-mode: # cat /proc/capi/controllers/1 name b1pciv4-a400 io