similar to: One way calling on asterisk to cisco call manager integration

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'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>

Hi All I am not sure what I wish to do it possible but I would like to see if you guys know any better. I have a site who has the extensions: 1231, 1232. 1233, 1234 Each of these users can dial each other on the extension number an also has an external CLI mapped to them. On all internal calls or calls to services such as call forwarding their Caller ID is: Name <XXXX> What

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

I have the option of running either SIP or SCCP for my cisco VoIP rollout..can someone shed light on what the pros/cons are? Seems everything is SIP these days so that's the option im leaning. Thanks- Matt

Oh Nooooooo! It looks as if there is a limit of 9 (not 10) maximum #include statments that you can have in extensions.conf. For example, the following extensions.conf causes the error to appear.. [user_3250071] #include "inc/wildwildwest/features.conf" #include "inc/wildwildwest/features.conf" #include "inc/wildwildwest/features.conf" #include

Issue is what subject says. Here is the background. Version: 11.11.0 Topology: Asterisk Box at our Data Center behind Cisco Firewall. Everything works fine from remote offices over a VPN. Issue is sales team would like to connect up to our Asterisk box remotely (offnet). Common enough solution, I'm guessing. So, I've opened all the correct holes on the firewall and hammered out

Actually now I have the phones with SIP firmware. I will try with 3pcc firmware along with XML files. Or any idea if we have CUCM application can we change the firmware. am ready to buy the developer edition. Regards . On Mon, 5 Dec 2016, 3:34 p.m. Steve Davies, <davies147 at gmail.com> wrote: > I tried... repeatedly... I failed. I bought some 3PCC phones, and they > just worked.

Has anyone tried to create a SIP trunk between Asterisk and a CUCM? If so has anyone enabled tls on the trunk? Would the tlscafile field in the Asterisk sip.conf be used to refer to the pem file provided by the CUCM? Is the purpose of tlscafile to refer to the other call manager's pem file? Or would the tlscafile field need to refer to the ca.crt file created for Asterisk using the asterisk

Hi All, I have a sip trunk up and running with a CUCM Express, passing calls fine except for a comfort noise error I'm getting on Asterisk: NOTICE[7520]: rtp.c:788 in process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: x.x.x.x I know Asterisk does not support comfort noise. I have "no comfort noise" on all

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've this infrastructure: |voip services| -- |*| -- |cme| -- |isdn| the voip services are logged on my *, then forwarded to number 601 on cme. The isdn calls too are forwarded to 601. On cme I've a timeout X for call-forward noan (no answer) to a specific number on * (5901) that is my x-lite software client. If 5901 is

Can't I upload the 3PCC firmware ? available from the Cisco website? Actually it came with sip88xx.... firmware. Regards . On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies147 at gmail.com> wrote: > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > they work very much like the Cisco SPA handsets. > > I also ended up with a non-3PCC

Boy, am I stuck... I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or maybe Asterisk just isn't mature enough yet. Nothing complicated really....

I have a SIP trunk built between a Cisco CallManager version 8. I can dial the phones registered to the Asterisk PBX from a phone registered to the Call Manager. I've tried to keep the config as small as possible to help the troubleshooting process. Attached is he most recent debug. My Callmanager IP address is 10.169.169.250, Asterisk server is 10.169.169.251 SIP.CONF [6001] type=friend

Kinda confused by this... I have a Cisco 7960 configured with a couple SIP extensions configured on the phone. Just trying to dial one extension from the other on the same phone, but when I do, I get: -- Remote UNIX connection -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack -- Called 2002 -- Got SIP response 486 "Busy here" back

Hi, after trying sshfs to mount a remote file system on a server with the result that sshfs will sooner or later get stuck and require a reboot of the client, I'm fed up with it and am looking for alternatives. So next I would like to use NFS over a VPN connection instead. To minimize the instances of the NFS mount getting stuck, it might be helpful to use autofs. What happens when the

Hi, I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is jitsi-----> asterisk server-----> analog PBX ----> landline phone I configured this scenario as follow in chan_dahdi.conf file ; General options [channels] usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes

Can someone tell me why the following from extensions.conf generates this error on asterisk load? Jan 13 09:27:24 WARNING[31701]: config.c:938 ast_config_internal_load: Maximum Include level (10) exceeded As far as I can tell I don't have a DEPTH of 10 includes. I certainly have more than 10 include statements, but I don't have them drilled down to a depth of 10. I'm thinking the

Hi, ? I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager). The connection seem to be there. When I do a "sip show peers" on the Asterisk server?I see the Callmanager as Monitored and online however I can't get any calls

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6