similar to: about timeouts

I'm trying to integrate my Asterisk box with my call manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2001] type=friend secret= dtmfmode=rfc2833

Dear, i using this scenario. jitsi---> asterisk---->EPABX------> Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88 at myphones:1] Dial("SIP/sandeep-00000004", "DAHDI/g0/88,20,rt") in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a

Hi, My rsync is stopped working suddenly I got following in verbose and log, mkstemp failed: No such file or directory and rsync error: received SIGUSR1 or SIGINT (code 20) at rsync.c(229) my rsync code : rsync -az -e ssh --delete $HOSTTOBACKUP:$SOURCE $DR_BACKUP_DIR/daily.0 >$tempfile 2>&1 the same code was working last week, what will be the problem, how to proceed to fix?

I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Mikel

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>

Hello list. No intention to start a flamewar here but I would really like opinions from those who know both the Cisco and Asterisk system. I'm working for a company with 15 offices in 11 countries, offices are relatively small (3-20 people each) and most of them have a Cisco 1760 Router installed with Call manager express (CME) and 1-3 ISDN lines (2-6 simultaneous calls). We

First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

Hello, I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front of the phone and also to hopefully resolve some issues with the phones not registering after a long period. Once we upgraded the phones now display "Error Verifying Config Info" in the Status messages and will not process the

I got this nice TDM22B with two green modules left and two modules right /var/log/messages shows: Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on major 196 Nov 27 00:37:53 dns kernel: Freshmaker version: 71 Nov 27 00:37:56 dns kernel: Freshmaker passed register test Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO Nov 27 00:37:56 dns kernel: Module 1: Installed

I got this nice TDM22B with two green modules left and two modules right /var/log/messages shows: Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on major 196 Nov 27 00:37:53 dns kernel: Freshmaker version: 71 Nov 27 00:37:56 dns kernel: Freshmaker passed register test Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO Nov 27 00:37:56 dns kernel: Module 1:

Hello List. I am setting up asterisk as a central dialplan, voicemail and conference solution, connected to 12 Cisco 1760 Routers running Call Manager Express IOS distributed around the world. This is all done over VPN. These routers all have PSTN access in their respective country. So far all is good, and Asterisks interopability with the Cisco CME using SIP is very good, although

Hi all, I'm porting a package that I've worked on for OS X to Cygwin/Windows. This package requires a Makefile. My question is, how can I find out (or what is), the link command? Here is the OS X Makefile: RLIB_LOC=${R_HOME} F90_FILES=\ class_data_frame.f90 \ class_old_dbest.f90 \ class_cm_data.f90 \ class_cm.f90 \ class_bgw.f90 \ class_cm_mle.f90 \ cme.f90 FORTRAN_FILES=\ dgletc.f

My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? [root at robin asterisk]# cat chan_dahdi.conf [trunkgroups] [channels] [phone](!) usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes

Hi all, I'm running Asterisk since two days, and it's really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me?

Hello All, We have been doing Asterisk and CME implementations recently but we almost always exlusively bring in analog lines and or PRI for PSTN access to our systems. I have known about providers providing SIP based lines and SIP trunks to end users for PSTN access. I am curious about the following: - How practical is this? The idea of terminating pstn calls to across the Internet