[asterisk-users] bash: asterisk: command not found
Hello Ive installed asterisk 14.2 on centos 6.8 but i am not able to start it below is what am executing and those are the errors anything am doing wrong? [root@localhost ~]# asterisk -r bash: asterisk: command not found [root@localhost ~]# asterisk -vvvc bash: asterisk: command not found chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 -Asterisk server - Analog lines
On 12/06/2016 06:45 PM, Motty Cruz wrote: Any suggestions on how to convert digital signal to analog? I do this with an ADIT 600 channel bank that I got off of ebaY many years ago. A quick search shows one for around $98 http://www.ebay.com/itm/like/191565965269?lpid=82&chn=ps&ul_noapp=true Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 -Asterisk server - Analog lines
Hello All, The problem I'm facing is the following: two machines that require analog signal. However, connection to the world is setup as follow: T1 connection to Asterisk server Any suggestions on how to convert digital signal to analog? Type of card on my Asterisk server is wct4xxp Thanks for your support! Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new inbound DID provider... no auth?
Hi, That's right - you just need to define a peer with a static IP address and "type=peer" to assign incoming calls to a peer name and apply the corresponding configuration (e.g. codecs). To make your configuration less redundant you can use templates in your peer definition (at least for chan_sip, I'm not sure if the same syntax applies on chan_pjsip). Example: --- ;; All configuration made to this peer will be applied to all childs of this definition [your-did-provider](!) type=peer allow=ulaw,alaw,g722 ... ;; This peer derives all other configuration from "your-did-provider", ;; then your local changes are applied and can override the derived ones. [your-did-provider-gw1](your-did-provider) host=1.2.3.4 [your-did-provider-gw2](your-did-provider) host=1.2.3.5 --- That's the shortest thing I can imagine at the moment. At least, with this way of definition you only need to do changes on one single point, not for every gateway IP. Am 30.11.2016 um 22:10 schrieb Jeff LaCoursiere: > > We are trying to work with a new DID provider and I find myself confused. > Their standard integration is to send the call with no authentication. I am > expected to whitelist all their possible gateways, and accept their calls I > guess with no peer definition. I actually have it working this way; the > calls land in our "public" context, I guess as "guest", and I am able to > route them from there. But that makes me nervous. > > I would rather at least have them be associated with a defined peer, so I can > set the right context and any other parameters I might want associated. It > is inbound only, no outbound. I might try to set a host= in a peer > definition with no secret, and see if that matches it, but I would rather > avoid making a peer definition for every gateway they have. Can anyone think > of a way to define a single peer that might show from multiple potential > addresses without authentication info? > > Cheers, > signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change Media IP in SDP
Hi, normally, Asterisk handles RTP IP addresses in SDP correctly, if you have specified - that NAT traversal is enabled for all peers (e.g. nat=force_rport,comedia) - your local network with "localnet=yournetwork/networkmask" - e.g. "localnet=192.168.1.0/255.255.255.0" - directmedia, canreinvite, directrtpsetup is deacitivated In this case, your Asterisk will always stay in the RTP stream and signalling only it's own IP address to other peers which is configured for the interface which Asterisk uses to reach the peer. But: If you have only one interface configured on your Asterisk server and an external firewall/router is managing your separated networks, this might not help. In this case you can use "externip" on a per peer basis in your SIP configuration to specify the IP address Asterisk uses in the SDP. Maybe, a global configuration of "externip" and "localnet" is all you need to help Asterisk setting the SDP address correctly. Also, enabling ICE support can help you getting the correct IP address if the remote peer supports it. Greetings Max Am 07.12.2016 um 00:02 schrieb Harel: > Hello List, > I need your help with information going out on my SDP. > Is it possible to update the Media Address on a per-call basis or a > per-channel basis? > Reason: > My Asterisk is in a private network and needs to connect to UA on its > internal network and also few external networks. One network is public and > the others are not public. Between each other the external networks are not > routable. Signaling is flowing with no issues because SIP Registers and NAT > boxes maintain sessions correctly. The problem is with RTP. After making > traces on all possible nodes of this network I clearly found out that the RTP > fails because the Asterisk doesn't manage to communicate the correct address > to the UAs in the SDP. It will report its internal IP address and the remote > UA will try to send its RTP to this address which, of course, will fail > miserably. > Obviously I can't use externaddr or media_address in sip.conf because it will > only be good for one network while the other external networks will fail just > the same. Same applies for STUN, it will only be good for the network the > STUN requests are being sent from. > On all networks I have fix IP addresses on my side and I fully control a > professional security box. > Asterisk is 13.6.0 > I can't, and don't want to, touch user-side equipment which is normally some > kind of voip phone behind a standard home VDSL router. > > Any ideas how can I transmit the correct IP address in SDP to UAs on > different networks? > > Many thanks, > Harel > > signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change Media IP in SDP
Hello List, I need your help with information going out on my SDP. Is it possible to update the Media Address on a per-call basis or a per-channel basis? Reason: My Asterisk is in a private network and needs to connect to UA on its internal network and also few external networks. One network is public and the others are not public. Between each other the external networks are not routable. Signaling is flowing with no issues because SIP Registers and NAT boxes maintain sessions correctly. The problem is with RTP. After making traces on all possible nodes of this network I clearly found out that the RTP fails because the Asterisk doesn't manage to communicate the correct address to the UAs in the SDP. It will report its internal IP address and the remote UA will try to send its RTP to this address which, of course, will fail miserably. Obviously I can't use externaddr or media_address in sip.conf because it will only be good for one network while the other external networks will fail just the same. Same applies for STUN, it will only be good for the network the STUN requests are being sent from. On all networks I have fix IP addresses on my side and I fully control a professional security box. Asterisk is 13.6.0 I can't, and don't want to, touch user-side equipment which is normally some kind of voip phone behind a standard home VDSL router. Any ideas how can I transmit the correct IP address in SDP to UAs on different networks? Many thanks, Harel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk install challenge
am new to asterisk and am trying to "make all". or dahdi install but in only reach this stage below on centos 6.8 . Any idea how to resolve or bypass this configure: creating ./config.status ./configure: line 18858: cannot create temp file for here-document: No such file or directory configure: error: write failure creating ./config.status make: *** [all] Error 1 Thanks, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK
Yes its possible mine im using 2pcs of Huwawie 173 USB dongle (one for each ISP provider) and connected to raspberry-pi. I can call out and recived call with IVR to forward to the extensio numbers. On Tue, 6 Dec 2016 at 10:36, christopher kamutumwa wrote: > Is it possible to have a simcard configured and become incoming line > > and outgoing on asterisk and also have the IVR function? If yes wat > > hardware is required to have this Accompished > > > > Thanks > > > > > > Chris > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK
Hi Chris On Tue, 2016-12-06 at 04:36 +0200, christopher kamutumwa wrote: > Is it possible to have a simcard configured and become incoming line > and outgoing on asterisk and also have the IVR function? Yes, it is possible! :-) A cheap solution is using a 3G-UBS-dongle. I have two SIM cards working in my Asterisk. I'm using two 3G-dongles (one for each SIM), a Huawei E173 (firmware 11.126.85.00.209) and a Huawei E180 (firmware 11.126.10.01.68). Google for "asterisk chan dongle" and you will find plenty of infos. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP 8841 asterisk integration
On 05/12/16 17:57, Gopalakrishnan N wrote: True agree, problem is somehow the people purchased am supporting to overcome that. Trying level best... around 20 phones has been purchased I have been able to use the following Cisco IP Phone with Asterisk. - Cisco SPA303, Cisco SPA504, Cisco 7941 and Cisco 8941 The SPA version just work as they are real SIP clients, however the IP range, 7xxx, 8xxx, 9xxx are SCCP clients, converting them as you did to SIP clients, is the Cisco SIP client and runs over TCP, not UDP as a normal SIP client. Coming back to your question, you will need to setup a DHCP with BOOTP and TFTP wit XML distribution, as you have done for the firmware distribution and Asterisk needs to have SIP wit TCP support. Additionnaly you need to generate a Cisco SIP XML configuration file and place it on your TFTP server with the MAC address. The directory and video I have never been able to get working, however as a SIP client, it works fine. I have these phones running with a number of customers in production and they all work, we actually mix types of phones, as not to require customers to purchase new IP phones, however sometimes the effort and time, which translates into cost, is too high. 😔😔 On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, mailto:mefhigos...@gmail.com>> wrote: With all the money you plan to invest in firmware, licenses, etc., you have bought a Grandstream IP phone or Yealink... Better still use a Raspberry Pi as a IP Softphone, or your Android phone with SIP client. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
