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Hello, I've got a Junghanns ZapHFC E1 PRI Card (cwain) and this driver writes very much messages into /var/log/messages like the following: --- snip --- Aug 2 17:58:02 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:02 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:02 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:12 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:12 asterisk1 k...

...ess here = 0, tmp->channel = 1, channel = 1 Mar 15 16:12:37 ERROR[31964] chan_zap.c: Unable to register channel '1' Mar 15 16:12:37 WARNING[31964] loader.c: chan_zap.so: load_module failed, returning -1 Mar 15 16:12:37 WARNING[31964] loader.c: Loading module chan_zap.so failed! [root@asterisk1 ~]# +++++++++++++++++++++++ END /var/log/asterisk/full++++++++ +++++++++++++++++++++++ /var/log/messages++++++++ [root@asterisk1 ~]# tail -20 /var/log/messages Mar 15 16:24:08 asterisk1 safe_asterisk: [pbx_config.so] => (Text Extension Configuration) Mar 15 16:24:08 asterisk1 safe_asterisk:...

Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28

I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvvvvvvvvvvvvvvvvvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is the same as current 0 gl_pathc 0...

I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running ztcfg and asterisk fails. [root@asterisk asterisk]# ztcfg Notice: Configuration file is /etc/zaptel.conf line 3: Unable to open master device '/dev/zap/ctl'

Hi Please help me understand about the below issue ? [root at asterisk1 ~]# /etc/init.d/asterisk restart Stopping safe_asterisk: [ OK ] Shutting down asterisk: [ OK ] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted...

Hi everybody, I'm testing asterisk@home 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from syslogd@asterisk1 at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user xxxx@yahoo.com could not be reached and body was like if it was the result of commands ifconfig etc unfortunally I don't have the message anymore but I went to the log and saw...

Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan,...

...sk2 ~> crm_mon -1 ============ Last updated: Thu Mar 28 06:09:54 2013 Stack: Heartbeat Current DC: asterisk2 (b966dfa2-5973-4dfc-96ba-b2d38319c174) - partition with quorum Version: 1.0.12-unknown 2 Nodes configured, unknown expected votes 1 Resources configured. ============ Online: [ asterisk1 asterisk2 ] Resource Group: group_1 asterisk_2 (lsb:asterisk): Started asterisk1 IPaddr_10_3_152_103 (ocf::heartbeat:IPaddr): Started asterisk1 Failed actions: p_asterisk_start_0 (node=asterisk1, call=64, rc=1, status=complete): unknown error p_asterisk_sta...

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the following message on asteriskm's cli:...

...ILED] STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1 Automatically restarting Asterisk Asterisk ended with exit status 1 Asteirsk died with code 1 Automatically restarting Asterisk ------------------------- Asterisk could not start! use tail etc Output was: [root@asterisk1 root]# tail /var/log/asterisk/full Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata_additional.con f': Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata_additional.conf': Found Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata-...

...am running into is that when the call to my cellphone is made, it appears as though the call "completes" so it never rolls to asterisk voicemail. Here is my current config: exten => 102,1,Dial(${sipura},10,) exten => 102,n,playback(pls-wait-connect-call) exten => 102,n,Dial(IAX2/asterisk1/9139275900,10,r) exten => 102,n,VoiceMail(u102@default) exten => 102,107,VoiceMail(b102@default) Here is the log from asterisk: -- Executing [102@internal:2] Playback("SIP/101-0a1178c0", "pls-wait-connect-call") in new stack -- Playing 'pls-wait-connect-call...

Hi guys I am currently getting the following in my log asterisk1*CLI> Parsing '/etc/asterisk/manager.conf': Found asterisk1*CLI> == Parsing '/etc/asterisk/manager_custom.conf': Found asterisk1*CLI> == Connect attempt from '127.0.0.1' unable to authenticate Can anyone tell me why asterisk would not be able to authentica...

Hi, Could anyone knows what went wrong with the error below result of installation of libsupertone. [root@asterisk1 latest]# tar xvf libsupertone-0.0.2.tar libsupertone-0.0.2/ libsupertone-0.0.2/AUTHORS libsupertone-0.0.2/Makefile.am libsupertone-0.0.2/COPYING libsupertone-0.0.2/config/ libsupertone-0.0.2/config/ltmain.sh libsupertone-0.0.2/config/missing libsupertone-0.0.2/config/install-sh libsupertone-0.0.2/c...

Hello all, For some reason I am not showing registration in SIP. Can anyone give me an idea what can cause this? asterisk1*CLI> sip show registry Host Username Refresh State -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/bd3a7577/attachment.htm

Has anyone else installed the official 1.4.0 release? I have, and it installed very easily. However, I don't have any of my usual command line tools for monitoring and debugging zap channels and PRI lines: asterisk1*CLI> pri show span 1 No such command 'pri show' (type 'help' for help) asterisk1*CLI> Ditto with zap stuff: asterisk1*CLI> zap show channels No such command 'zap show' (type 'help' for help) asterisk1*CLI> I didn't see these in the...

Hi Please help me understand about the below issue ? [root at asterisk1 ~]# /etc/init.d/asterisk restart Stopping safe_asterisk: [ OK ] Shutting down asterisk: [ OK ] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted...

Hi, I need some information on DID and Callback. Please read-on: Question on DID (User1 Calling User2 via normal Telephone line and sending its CLI: Connectivity is as below: User1 ==PSTN==> DigiumE1/Asterisk1 ==INTERNET==> DigiumE1/Asterisk2 ==PSTN==> User2 1. Can User1 make a single stage call to User2 via Asterisk1? Currently User1 is able call User2 on Two Stage basis (Asterisk answers, and then user1 hears a message, and then he is allowed to call) Question on CALLBACK. Connectivity diagr...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > asterisk2 server Now, let me explain the symptoms. d-channel errors on the asterisk1 server on span1 (which is the line coming from the echo cancel from the ISP). as...

...I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk as followed: [SIP_BD1] type=peer qualify=yes host=192.168.0.254 disallow=all context=from-pstn allow=h723 and inside the quantum I change the config sip useragent to 5060. Up to this part if I run sip show peers, I got: asterisk1*CLI> sip show peers Name/username????????????? Host??????????? Dyn Nat ACL Port???? Status SIP_BD1??????????????????? 192.168.0.254?????????????? 5060???? OK (56 ms) Which seems that I can connect to the quantum A800, but when ever I tried to call I can?t get the phone connected. I mean the des...