Re: [asterisk-users] Call queues on load-balanced asterisks
Hello Mr. Liu, I tried searching for more information about FlexQueue, where to download etc. Google linked to vicidial.cn, which appears in your signature, but that page is all in chinese, and I couldn't find any english link. Where can I get more information about it? Is it a commercial product? With kind regards, Pan B. Christensen Ibidium AS http://www.ibidium.no - Original Message - From: Thomas Liu thomas@wshuttle.com To: asterisk-users@lists.digium.com Sent: Tuesday, January 11, 2011 5:14 PM Subject: Re: [asterisk-users] Call queues on load-balanced asterisks Hi Pan Dhaval, We have implemented a FastAGI based queue with Erlang for a inbound call center, and call this new application as FlexQueue. All calls distributed on multiple asterisk boxes go through and are controlled by that same remote fastagi server. It can routing calls to any destination, by any business rules. It don't rely on the db for agent/call status store query. It's event driven and dict based agent/call store query, with very good performance, and low cpu power consumption. I think for your requirement, app_queue could not fulfill that. Best Regards, Thomas Liu - WShuttle Infotech Ltd. http://www.wshuttle.com / http://www.lookmypc.com http://www.vicidial.cn / http://www.call-center-software.com.cn Tel: +86 20 39230098 39230096 Mobile : +86 1390 3051 930 HK DID: +852 6950 0916, Macau DID: +853 6285 0645 Email: thomas@wshuttle.com MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly Yahoo Messenger: thomaslly Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, Guangzhou Higher Education Mega Center, Guangzhou, Guangdong Province, China. Zip code: 510006 -- Hello Dhaval (and others), As far as I can tell, realtime queue will not solve my problem. I can statically define the same queue with the same members on two machines as well. I was planning to use realtime anyway. The issue is the actual queueing of the incoming calls. Let?s say I define the queue IT-support with members Local/100 and Local/101 on both machines. The first call comes in and is distributed by Kamailio to Asterisk A, and answered by 100. The next call comes in to Asterisk B, and is answered by 101. At this point, both members are busy. Call 3 now comes in and is sent to Asterisk A, where it waits for a free member. Call 4 comes in and is also sent to Asterisk A, as is Call 5. Then call 6 is sent to Asterisk B. At this point 100 finishes his call and becomes free. Which call is delivered to 100? As far as I can tell, that?s a 50/50 chance between call 3 and call 6. This is not correct behaviour! Call 6 should wait until calls 3, 4 and 5 (from the other server) have all been delivered. In the example above: When call 3 comes in, Asterisk A may even try to deliver it to 101, who gets call waiting indication. He will now have two simultaneous calls from the same queue! I have not found any way to share information about calls waiting in the queue, wait times, member states and so on between the two servers. Unless you guys know of a way, I think I'm going to have to ask the customer to change their design to master-slave (with failover) instead of load-balanced. With kind regards, Pan Hello Pan, You can user DB for this just make real time configuration of Queue and make all asterisk server connected to Same DB if more load then use replication for different server on DB, also So that Quque name should be same for all server and asterisk can call same agent. you didnot mentioned that which purpose youwere use queue other wise i can give answer in better way. regards Dhaval On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen pan at ibidium.no wrote: Hello, I have been asked to implement the following design: Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question. All this is working fine. I've now been asked to start implementing calling queues, and my question is this: How can I implement the same queue on multiple Asterisk servers? Let's say that 10 people call the same queue. These calls would then currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A respect the 5 people queued on the other server and vice versa? Will the customer need to change their design to make the feature servers master-slave with failover instead of load-balanced? Mvh Pan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Re: [asterisk-users] SetVar Warning
Steve Edwards wrote: I don't have a 1.4 system on hand, but 1.2 1.6 use set(). 1.4.x uses Set() as well. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING T.30 ECM carrier not found
Hi list, I have search for a clear explanation about this mensage WARNING T.30 ECM carrier not found, but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING T.30 ECM carrier not found
CORRECTING: I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXS dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 13 Jan 2011 09:51:24 -0200 Subject: [asterisk-users] WARNING T.30 ECM carrier not found Hi list, I have search for a clear explanation about this mensage WARNING T.30 ECM carrier not found, but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to get Fax t38 working with IrisTel trunk
On 01/11/2011 04:28 PM, Karim Mardhani wrote: Hi everyone, I have been trying to get T.38 Faxing to work with Iristel sip trunks for last few days but havn't been sccussful. I am using Asterisk 1.6.2.8 and SpanDSP 0.6. Here is what I see in the tcpdump capture: You are 7 versions behind on Asterisk (the current version is 1.6.2.15, and 1.6.2.16 will be released today or tomorrow), and there were significant improvements in T.38 negotiation between those releases. Before spending too much more time on this, I would strongly suggest you update to the latest release in the 1.6.2 series. 1. Call come in from the trunk as regular voice call with g.711 codec 2. Asterisk answers the call and recognizes the CNG and sends the call to fax extension 3. Eventually Receive fax is called with a file name 4. Asterisk sends update message to remote gateway with T38 codec information Do you mean a SIP 'UPDATE' request? 5. Remote server doesn't respond. Asterisk resends update messages multiple times. Then the IrisTel SIP implementation is broken. If it receives a SIP request it does not understand, or does not want to process, it *still* has to respond to indicate that to the sender of the request. Ignoring the request will result in broken behavior. 6. Eventually remote gateway sends the invite with T38 codecs listed in the SDP I assume you mean a re-INVITE here. 7. Asterisk Responds back with 488 Not acceptable here Was ReceiveFAX still running when this happened, or had it given up trying to switch the session to T.38 mode? Asterisk will respond with 488 to a T.38 re-INVITE when it cannot switch the channel to T.38 mode because there isn't a T.38 endpoint available... and ReceiveFAX will only wait for a reasonable period of time for the sending endpoint to switch to T.38 before falling back to audio FAX mode. 8. Another invite is send from the remote gateway with T38 codecs in the SDP Was image/t38 the *only* media format offered? 9. Asterisk sends OK but g.711 codecs listed in SDP. 10. remote gateway sends the BYE message and call is completed. The questions I have are: Why Asterisk sends update message in step 4 above instead of send an invite? Do you have 'directmedia=update' or 'canreinvite=update' set in sip.conf for this endpoint? If so, Asterisk will send UPDATE instead of re-INVITE requests because you told it to :-) Why Asterisk responds back with 488 in step 7 above? Why asterisk sends g.711 codecs in step 9 above? This will depend on the answers to the questions I included above. When debugging problems like this, it is very important to have both a packet capture *and* a complete console log (with as much verbosity and debugging enabled as possible) so that all factors involved can be considered. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
On 01/11/2011 02:20 PM, Gilles wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. By any chance, would someone have a working configuration so I can take a look? I have had OpenVPN and Asterisk running together, with both Linux and Windows clients for about 2 years now. As the others have already pointed out - you really should sort out your OpenVPN setup first and basic connectivity first between client and server - before starting to wonder what is happening on the Asterisk side of things. I'm not sure what book you read on OpenVPN - but if you read carefully and follow the step by step instructions on OpenVPN.net (under Community Edition) - you shouldn't have too much trouble getting a basic, certificates based connection working. I am a subscriber to the openvpn users mailing list as well - and if you follow the steps and still get stuck - feel free to post there what you've done so far and what is not working - and I'm sure people will be happy to point you in the right direction. Incidently - I use QuteCom (qutecom.org) and X-lite on Windows, and Linphone, Twinkle and Ekiga on Linux as clients - if it helps. Sebastian Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax stopped working when upgrading to 1.8.2
Gentlemen, We have a setup as below: PSTN – E1 – Avaya – OOH323 trunk – Asterisk – SPA-2102 – Fax machine Running Asterisk SVN-trunk-r280589M, fax working as a clock. I decided to leave “trunk” and go a stable version so I upgraded to 1.8.2. Didn’t change any config files, everything worked as before except fax. I wonder if there are any known issues or things that I have missed to do in some config file. Did a downgrade to SVN-trunk-r280589M and fax started to work again. /Magnus-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
On Tue, 11 Jan 2011, Gilles wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting through Ethernet in hotels or public wifi hotspots. Are you sure that your uClinux system can actually handle the load of the VPN encryption? If it doesn't have a hardware crypto chip then I'd be very wary of it. And you might just want to make your own life easier - and not use the appliance for the VPN endpoint, but something else - e.g. Draytek 2820 (or a 2900 equivalent) and then the clients can use bog-standard Microsoft pptp. You might save yourself a whole load of grief that way. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
On Tue, 11 Jan 2011 15:20:39 +0100, Gilles codecompl...@free.fr wrote: By any chance, would someone have a working configuration so I can take a look? Got it working :-) Thanks much guys for the help. For those interested, here's how I did it. Note that the appliance only has the openvpn server, so I used a Ubuntu workstation to create the certificates + keys: = 1. Install OpenVPN on Asterisk server. On appliance, there's only a single binary /bin/openvpn, and configuration files are in /etc/openvpn/. To be positive SIP/RTP packets go through the OpenVPN tunnel, make sure the firewall in front of the OpenVPN/Asterisk server only has OpenVPN port open (default: UDP 1194). 2. On client, from www.openvpn.net, download and install OpenVPN for Windows, which includes Service + GUI 3. If using an appliance with just the openvpn binary, use a workstation to install the OpenVPN package and create certificates + keys: apt-get install openvpn 4. On workstation, copy programs to create keys and certificates: mkdir /etc/openvpn/easy-rsa cp -R /usr/share/doc/openvpn/examples/easy-rsa/2.0/* /etc/openvpn/easy-rsa 5. Create the CA, and one pair of public/private keys for each host (server, clients) #Always use a unique Common Name vi /etc/openvpn/easy-rsa/vars #export variables . ./vars ./clean-all ./build-ca ./build-dh #keys for server ./build-key-server server #keys for client ./build-key client1 6. Create configuration file for server /var/www/server.ovpn: port 1194 proto udp dev tun ca ca.crt cert server.crt key server.key dh dh1024.pem #server will use this network number for OpenVPN tunnel, server = 10.8.0.1 server 10.8.0.0 255.255.255.0 ifconfig-pool-persist ipp.txt keepalive 10 120 #Uncomment if compiled with compression #comp-lzo persist-key persist-tun status openvpn-status.log verb 3 7. Create configuration file for client /var/www/client1.ovpn: dev tun proto udp remote public IP to reach OpenVPN/Asterisk server 1194 resolv-retry infinite nobind persist-key persist-tun ca ca.crt cert client1.crt key client1.key #comp-lzo verb 3 8. Copy keys/certificates/config files to www so can be downloaded by server and client cd /etc/openvpn/easy-rsa/keys cp ca.crt dh1024.pem server.crt server.key client1.crt client1.key server.ovpn client1.ovpn /var/www #So web server can send files chmod 644 /var/www/server.key chmod 644 /var/www/client1.key 9. On server, download files: Asterisk cd /etc/openvpn Asterisk wget http://workstation/ca.crt Asterisk wget http://workstation/dh1024.pem Asterisk wget http://workstation/server.crt Asterisk wget http://workstation/server.key Asterisk chmod 600 server.key Asterisk wget http://workstation/server.ovpn 10. On client, download files: cd c:\program files\openvpn\config wget http://workstation/ca.crt wget http://workstation/client1.crt wget http://workstation/client1.key wget http://workstation/client.ovpn Launch server: Asterisk /bin/openvpn /etc/openvpn/server.ovpn Launch client: Start OpenVPN Service Start OpenVPN GUI with Admin rights: Right-click on OpenVPN GUI icon Connect ping 10.8.0.1 If ping OK, configure SIP client to connect to Asterisk through the server's private IP used by OpenVPN tunnel, eg. 10.8.0.1, and make a call. = HTH, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also make sure you have your externip setup as well. Else you will notice one way audio or cut off after 30 seconds. Rest of your work is all good. For security reasons the workstation that creates the keys is not connected to any network (local or internet) -Bruce On Thu, Jan 13, 2011 at 8:24 AM, Gilles codecompl...@free.fr wrote: On Tue, 11 Jan 2011 15:20:39 +0100, Gilles codecompl...@free.fr wrote: By any chance, would someone have a working configuration so I can take a look? Got it working :-) Thanks much guys for the help. For those interested, here's how I did it. Note that the appliance only has the openvpn server, so I used a Ubuntu workstation to create the certificates + keys: = 1. Install OpenVPN on Asterisk server. On appliance, there's only a single binary /bin/openvpn, and configuration files are in /etc/openvpn/. To be positive SIP/RTP packets go through the OpenVPN tunnel, make sure the firewall in front of the OpenVPN/Asterisk server only has OpenVPN port open (default: UDP 1194). 2. On client, from www.openvpn.net, download and install OpenVPN for Windows, which includes Service + GUI 3. If using an appliance with just the openvpn binary, use a workstation to install the OpenVPN package and create certificates + keys: apt-get install openvpn 4. On workstation, copy programs to create keys and certificates: mkdir /etc/openvpn/easy-rsa cp -R /usr/share/doc/openvpn/examples/easy-rsa/2.0/* /etc/openvpn/easy-rsa 5. Create the CA, and one pair of public/private keys for each host (server, clients) #Always use a unique Common Name vi /etc/openvpn/easy-rsa/vars #export variables . ./vars ./clean-all ./build-ca ./build-dh #keys for server ./build-key-server server #keys for client ./build-key client1 6. Create configuration file for server /var/www/server.ovpn: port 1194 proto udp dev tun ca ca.crt cert server.crt key server.key dh dh1024.pem #server will use this network number for OpenVPN tunnel, server = 10.8.0.1 server 10.8.0.0 255.255.255.0 ifconfig-pool-persist ipp.txt keepalive 10 120 #Uncomment if compiled with compression #comp-lzo persist-key persist-tun status openvpn-status.log verb 3 7. Create configuration file for client /var/www/client1.ovpn: dev tun proto udp remote public IP to reach OpenVPN/Asterisk server 1194 resolv-retry infinite nobind persist-key persist-tun ca ca.crt cert client1.crt key client1.key #comp-lzo verb 3 8. Copy keys/certificates/config files to www so can be downloaded by server and client cd /etc/openvpn/easy-rsa/keys cp ca.crt dh1024.pem server.crt server.key client1.crt client1.key server.ovpn client1.ovpn /var/www #So web server can send files chmod 644 /var/www/server.key chmod 644 /var/www/client1.key 9. On server, download files: Asterisk cd /etc/openvpn Asterisk wget http://workstation/ca.crt Asterisk wget http://workstation/dh1024.pem Asterisk wget http://workstation/server.crt Asterisk wget http://workstation/server.key Asterisk chmod 600 server.key Asterisk wget http://workstation/server.ovpn 10. On client, download files: cd c:\program files\openvpn\config wget http://workstation/ca.crt wget http://workstation/client1.crt wget http://workstation/client1.key wget http://workstation/client.ovpn Launch server: Asterisk /bin/openvpn /etc/openvpn/server.ovpn Launch client: Start OpenVPN Service Start OpenVPN GUI with Admin rights: Right-click on OpenVPN GUI icon Connect ping 10.8.0.1 If ping OK, configure SIP client to connect to Asterisk through the server's private IP used by OpenVPN tunnel, eg. 10.8.0.1, and make a call. = HTH, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
On Thu, 13 Jan 2011 09:43:26 -0500, Bruce B bruceb...@gmail.com wrote: In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also make sure you have your externip setup as well. Else you will notice one way audio or cut off after 30 seconds. I don't have sip_nat.conf, as I don't use any GUI to configure Asterisk. I didn't have to change anything to Asterisk as compared to when connecting directly. Since the other extensions live in the same LAN as Asterisk, should I configure localnet just for the remote extension that connects in through OpenVPN, while leaving 192.168.0.0/24 for the local extensions? The only issue I notice, is that Asterisk doesn't tell the other end when the local end has hung up, so the other end either remains online or hangs up after 20-30 seconds. I've tried XLite and ZoIPer, same result. This never happens when not going through the VPN. Has someone seen this? Here's the error message: -- Executing [siemens@internal:1] Dial(SIP/remote-00d22b1c, SIP/siemens) in new stack -- Called siemens -- SIP/siemens-00d329ec is ringing -- SIP/siemens-00d329ec answered SIP/remote-00d22b1c -- Packet2Packet bridging SIP/remote-00d22b1c and SIP/siemens-00d329ec == Spawn extension (internal,siemens, 1) exited non-zero on 'SIP/remote-00d22b1c' WARNING[82]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission NWQ2NTRhMzYxZjIzZTBhODY3NTBhYzMxMTk5MTUyYjY. for seqno 2 (Critical Response) Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
On Thu, 13 Jan 2011 15:55:10 +0100, Gilles codecompl...@free.fr wrote: The only issue I notice, is that Asterisk doesn't tell the other end when the local end has hung up, so the other end either remains online or hangs up after 20-30 seconds. Found it: We must add a localnet directive so that Asterisk hangs up the call OK: externip=public IP #local end-points localnet=192.168.0.0/255.255.255.0 #remote end-points through VPN localnet=10.0.0.0/255.0.0.0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Blf / Directed Pickup
Is the buddy watch tag activated in your mac-directory.xml file ? bw1/bw item lbSebastien/lb fnSebastien/fn lnThomas/ln ct222/ct sd1/sd bw1/bw /item --- Sebastien Thomas Amplisys Inc. - Digital Telephony Integration Specialists T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS On 2011-01-13, at 1:32 AM, Mark Murawski wrote: Would anyone happen to have some examples of polycom configs, specifically the 650 with sidecar for blf. I have the asterisk side all configured since I've set up blf with other types of phones, but I'm missing the polycom side. I've put together a mac-directory.xml, and the sidecar now lists numbers as speed dials but does not subscribe to blf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Blf / Directed Pickup
Yeah... My directory looks like this: directory item_list item ln6288/lnfn/fnct6288/ctsd1/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6208/lnfn/fnct6208/ctsd2/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6234/lnfn/fnct6234/ctsd3/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6205/lnfn/fnct6205/ctsd4/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6231/lnfn/fnct6231/ctsd5/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item /item_list /directory On 01/13/2011 10:20 AM, Sebastien Thomas wrote: Is the buddy watch tag activated in yourmac-directory.xml file ?bw1/bw item lbSebastien/lb fnSebastien/fn lnThomas/ln ct222/ct sd1/sd bw1/bw /item --- Sebastien Thomas Amplisys Inc. - Digital Telephony Integration Specialists T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS On 2011-01-13, at 1:32 AM, Mark Murawski wrote: Would anyone happen to have some examples of polycom configs, specifically the 650 with sidecar for blf. I have the asterisk side all configured since I've set up blf with other types of phones, but I'm missing the polycom side. I've put together amac-directory.xml, and the sidecar now lists numbers as speed dials but does not subscribe to blf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
As I said, your tunnel address should be part of localnet. Otherwise you experience what you did. -Bruce On Thu, Jan 13, 2011 at 10:00 AM, Gilles codecompl...@free.fr wrote: On Thu, 13 Jan 2011 15:55:10 +0100, Gilles codecompl...@free.fr wrote: The only issue I notice, is that Asterisk doesn't tell the other end when the local end has hung up, so the other end either remains online or hangs up after 20-30 seconds. Found it: We must add a localnet directive so that Asterisk hangs up the call OK: externip=public IP #local end-points localnet=192.168.0.0/255.255.255.0 #remote end-points through VPN localnet=10.0.0.0/255.0.0.0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Blf / Directed Pickup
Ok, that looks good. We use FreePBX, and I know I had to modify a couple Asterisk files to get the BLF working ... here are some of my mods but may also be used for FOP2 (I dont recall which go for BLF and which go FOP2). vi /etc/asterisk/sip_registrations_custom.conf allowsubscribe=yes vi /etc/asterisk/sip_custom.conf callevents=yes notifyringing=yes limitonpeers=yes I also override some of the sip.cfg settings in the polycom dir with: feature feature.1.enabled=1 feature.9.enabled=0 feature.18.enabled=1 / pres pres.reg=1 pres.idleSoftkeys=0 / --- Sebastien Thomas Amplisys Inc. - Digital Telephony Integration Specialists T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS *** Need help? Contact supp...@amplisys.ca *** On 2011-01-13, at 10:29 AM, Mark Murawski wrote: Yeah... My directory looks like this: directory item_list item ln6288/lnfn/fnct6288/ctsd1/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6208/lnfn/fnct6208/ctsd2/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6234/lnfn/fnct6234/ctsd3/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6205/lnfn/fnct6205/ctsd4/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6231/lnfn/fnct6231/ctsd5/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item /item_list /directory On 01/13/2011 10:20 AM, Sebastien Thomas wrote: Is the buddy watch tag activated in yourmac-directory.xml file ?bw1/bw item lbSebastien/lb fnSebastien/fn lnThomas/ln ct222/ct sd1/sd bw1/bw /item --- Sebastien Thomas Amplisys Inc. - Digital Telephony Integration Specialists T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS On 2011-01-13, at 1:32 AM, Mark Murawski wrote: Would anyone happen to have some examples of polycom configs, specifically the 650 with sidecar for blf. I have the asterisk side all configured since I've set up blf with other types of phones, but I'm missing the polycom side. I've put together amac-directory.xml, and the sidecar now lists numbers as speed dials but does not subscribe to blf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN + SIP configuration?
On Thu, 13 Jan 2011 10:42:48 -0500, Bruce B bruceb...@gmail.com wrote: As I said, your tunnel address should be part of localnet. Otherwise you experience what you did. Sorry about that. I didn't make long-enough calls for Asterisk to disconnect due to the lack of localnet for the VPN, and didn't know we could have multiple localnet directives. Thanks for pointing it out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2
Magnus, Can it be the same as I experienced https://issues.asterisk.org/view.php?id=18542 ? Do not be confused by the ticket subject, it reflects the symptoms as they looked originally You can try the patch if applicable and if you know how to compile Addons in 1.8 separately or if you have a capacity to compile the whole thing. -Vladimir On 1/13/2011 6:31 AM, magnu...@inputinterior.se wrote: Gentlemen, We have a setup as below: PSTN – E1 – Avaya – OOH323 trunk – Asterisk – SPA-2102 – Fax machine Running Asterisk SVN-trunk-r280589M, fax working as a clock. I decided to leave “trunk” and go a stable version so I upgraded to 1.8.2. Didn’t change any config files, everything worked as before except fax. I wonder if there are any known issues or things that I have missed to do in some config file. Did a downgrade to SVN-trunk-r280589M and fax started to work again. /Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue_log in MySQL database
Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_log in MySQL database
Hi Jonas, On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... I don't think Asterisk has this support built-in...maybe 1.8 does? However, what I do to manage queue_log is I have a small daemon that I have written in Python that watches the queue_log file, parses each incoming line, and stores it in a MySQL table. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_log in MySQL database
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, January 13, 2011 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] queue_log in MySQL database Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... Kind regards, Jonas. I'd say that depends on your release. Check this link http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_log in MySQL database
On 01/13/2011 05:25 PM, James Lamanna wrote: Hi Jonas, On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellensjonas.kell...@telenet.be wrote: Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... I don't think Asterisk has this support built-in...maybe 1.8 does? However, what I do to manage queue_log is I have a small daemon that I have written in Python that watches the queue_log file, parses each incoming line, and stores it in a MySQL table. -- James -- I actually found this : http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL But a second question : how can I know how long a caller stayed inside the queue untill it was answered by a member ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF not being heard correctly by far end conference system
Hi Duncan, On Wed, Jan 12, 2011 at 10:13 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi Thorsten Thanks very much, at this point my preference is rfc2833 but I will try some other options. The system is generating audible tones (that I can hear), although I think the audio is generated by the last sip device in the network so if thats so I don't have any control of it. Probably then I have to go to inband to get some control back, I am not sure what I lose from this, or change upstream provider (although the current provider works from a different system) In my DTMF experience I have found a few IVRs and conference systems out there that won't accept my DTMF, even though its DTMF that I can see going out over PRI channels. My guess is that these systems use too tight of a duration window on their DTMF detectors. In your case I'm guessing that for some reason the SIP DTMF tones are coming out with too short of a duration. I believe you can fiddle with the dtmf tone duration and spacing in channel.c but I don't know if that will fix the issue. Is it possible to get the DTMF specs from the manufacturer of the conference system? -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_log in MySQL database
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, January 13, 2011 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] queue_log in MySQL database On 01/13/2011 05:25 PM, James Lamanna wrote: Hi Jonas, On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellensjonas.kell...@telenet.be wrote: Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... I don't think Asterisk has this support built-in...maybe 1.8 does? However, what I do to manage queue_log is I have a small daemon that I have written in Python that watches the queue_log file, parses each incoming line, and stores it in a MySQL table. -- James -- I actually found this : http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL But a second question : how can I know how long a caller stayed inside the queue untill it was answered by a member ?? Kind regards, Jonas. Just a WAG, but I'm guessing that you would cross-reference to the CDR for this information. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID and URL pop up for windows...
Anyone has a good recommendation for a Windows program that will open a browser URL when your phone receives a call? We had been using Yaacid but since it is no longer being developed we need to look for an alternative. It should be light weight and work on all versions of Windows. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_log in MySQL database
On 13 January 2011 16:28, Jonas Kellens jonas.kell...@telenet.be wrote: I actually found this : http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL But a second question : how can I know how long a caller stayed inside the queue untill it was answered by a member ?? The queue_log table contains exactly that information - Along with a few other events, it indicates when a caller joined a queue, and when an agent gets given the call. Take the difference between the 2 times and you have the number that you need. Cheers, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID and URL pop up for windows...
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Thursday, January 13, 2011 11:37 AM To: Asterisk Subject: [asterisk-users] CallerID and URL pop up for windows... Anyone has a good recommendation for a Windows program that will open a browser URL when your phone receives a call? We had been using Yaacid but since it is no longer being developed we need to look for an alternative. It should be light weight and work on all versions of Windows. This might be useful (or not) http://articleresource.org/internet-and-businesses-online/web-hosting/use-de sktop-pop-up-application-with-asterisk-pbx-111454 Unless you need a canned app, this would be an easy program to develop on your own. The easiest way (IMO) to do this would be to put a small instance of Apache on your Asterisk server and run a CGI program that interfaces to the local instance of Asterisk and pops a new window when a call comes in. This would have the added benefit of being self-contained on the Asterisk machine (no programs to install or ports to open). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_log in MySQL database
On 01/13/2011 11:25 AM, Danny Nicholas wrote: *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, January 13, 2011 10:19 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] queue_log in MySQL database Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... Kind regards, Jonas. I’d say that depends on your release. Check this link http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL Specifically, you're looking for the part I added that mentions the changes to how extconfig.conf entries are referenced. You need to use the context name, not the database name. You'll also want to note the information about changes made to the data structures for Asterisk 1.8. As far as your request about tracking the time a call is in the queue, that's information that is directly available in the queue_log. One important question that you haven't asked is How do I track how long each user was logged in to the queue, even if they received no calls?. That will require additions to your login/logout context that write entries to the log each and every time a user logs in/out. You can then report on that data. You might want to reconsider reinventing the wheel on this one. Have you checked into Queuemetrics at http://www.queuemetrics.com ? Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_log in MySQL database
On 01/13/2011 2:07 PM, Tom Rymes wrote: That will require additions to your login/logout context that write entries to the log each and every time a user logs in/out. You can then report on that data. While there's a thread going on about this topic, and while I've written the above comment, can anyone confirm that the QueueLog command will indeed write entries out to the realtime queue_log, not just the file based log? Many thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID and URL pop up for windows...
On Thu, 13 Jan 2011 13:06:36 -0600, Danny Nicholas da...@debsinc.com wrote: Unless you need a canned app, this would be an easy program to develop on your own. The easiest way (IMO) to do this would be to put a small instance of Apache on your Asterisk server and run a CGI program that interfaces to the local instance of Asterisk and pops a new window when a call comes in. What about a single-EXE Windows app that would connect to the Asterisk Manager Interface and display CID information when a call comes in? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID and URL pop up for windows...
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, January 13, 2011 4:14 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CallerID and URL pop up for windows... On Thu, 13 Jan 2011 13:06:36 -0600, Danny Nicholas da...@debsinc.com wrote: Unless you need a canned app, this would be an easy program to develop on your own. The easiest way (IMO) to do this would be to put a small instance of Apache on your Asterisk server and run a CGI program that interfaces to the local instance of Asterisk and pops a new window when a call comes in. What about a single-EXE Windows app that would connect to the Asterisk Manager Interface and display CID information when a call comes in? Not a bad idea, but possibly a security hole in that the AMI password would have to be imbedded in the application. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with ZAP Channels
Run asterisk in verbose and and dial zap. Make sure you have hangup dialplan. -- Sent from my iPhone On Jan 12, 2011, at 1:23 PM, Antonio Modesto mode...@isimples.com.br wrote: Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continues in use, even after hanging the call up, then i need to run the soft hangup Zap/zapchannel in the asterisk CLI to release the channel. Here is my zapata.conf: [trunkgroups] [channels] language=pt_BR context=default usecallerid=yes hidecallerid=no callwaiting = yes usecallingpres= yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=yes ;echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived loglevel=255 hanguponswitchpolarity=yes context=disc-from-trunk-ZAP001 pulsedial=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 busydetect=no busycount=5 callprogress=no cidsignalling=dtmf relaxdtmf=yes cidstart=polarity channel=1 Does anyone know what can i do to solve this problem? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID and URL pop up for windows...
What you need already exists: http://bestof.nerdvittles.com/applications/screenpop/ http://bestof.nerdvittles.com/applications/screenpop/But better thing would be to a have TAPI for outlook to query Outlook contact as well because it allows for making notes on the contact. I am willing to pay for that if it is added to URANG II -Bruce On Thu, Jan 13, 2011 at 5:30 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, January 13, 2011 4:14 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CallerID and URL pop up for windows... On Thu, 13 Jan 2011 13:06:36 -0600, Danny Nicholas da...@debsinc.com wrote: Unless you need a canned app, this would be an easy program to develop on your own. The easiest way (IMO) to do this would be to put a small instance of Apache on your Asterisk server and run a CGI program that interfaces to the local instance of Asterisk and pops a new window when a call comes in. What about a single-EXE Windows app that would connect to the Asterisk Manager Interface and display CID information when a call comes in? Not a bad idea, but possibly a security hole in that the AMI password would have to be imbedded in the application. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Blf / Directed Pickup
Thanks! Blf is working now. I forgot I had to set set subscribecontext. When a phone is ringing, the blf light is solid red and the icon is a (/) type icon indicating unavailable. I'm also interested in directed pickup. I set up the following: call.directedCallPickupString=*6 call.directedCallPickupMethod=legacy Hitting the button next to the contact will speed dial the contact instead of pick up the ringing call. On 01/13/2011 10:54 AM, Sebastien Thomas wrote: Ok, that looks good. We use FreePBX, and I know I had to modify a couple Asterisk files to get the BLF working ... here are some of my mods but may also be used for FOP2 (I dont recall which go for BLF and which go FOP2). vi /etc/asterisk/sip_registrations_custom.conf allowsubscribe=yes vi /etc/asterisk/sip_custom.conf callevents=yes notifyringing=yes limitonpeers=yes I also override some of the sip.cfg settings in the polycom dir with: feature feature.1.enabled=1 feature.9.enabled=0 feature.18.enabled=1 / pres pres.reg=1 pres.idleSoftkeys=0 / --- Sebastien Thomas Amplisys Inc. - Digital Telephony Integration Specialists T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS *** Need help? Contact supp...@amplisys.ca mailto:supp...@amplisys.ca *** On 2011-01-13, at 10:29 AM, Mark Murawski wrote: Yeah... My directory looks like this: directory item_list item ln6288/lnfn/fnct6288/ctsd1/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6208/lnfn/fnct6208/ctsd2/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6234/lnfn/fnct6234/ctsd3/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6205/lnfn/fnct6205/ctsd4/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item item ln6231/lnfn/fnct6231/ctsd5/sdrt2/rtdc/ad0/adar0/arbw1/bwbb0/bb /item /item_list /directory On 01/13/2011 10:20 AM, Sebastien Thomas wrote: Is the buddy watch tag activated in yourmac-directory.xml file ?bw1/bw item lbSebastien/lb fnSebastien/fn lnThomas/ln ct222/ct sd1/sd bw1/bw /item --- Sebastien Thomas Amplisys Inc. - Digital Telephony Integration Specialists T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS On 2011-01-13, at 1:32 AM, Mark Murawski wrote: Would anyone happen to have some examples of polycom configs, specifically the 650 with sidecar for blf. I have the asterisk side all configured since I've set up blf with other types of phones, but I'm missing the polycom side. I've put together amac-directory.xml, and the sidecar now lists numbers as speed dials but does not subscribe to blf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 5-7 second delay in connecting outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has answered the call. I've had people hang-up since they don't hear anything when they answer my calls. If I try the exact same call using an IAX route, the call is connected at my end just as soon as the PSTN number answers. I don't have any connection delays for incoming FXO calls. They are connected as soon as I answer the calls. Can anyone give me some pointers on where to start looking? Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2
Did apply the patch and did a recompile, no difference, fax still not working. But I did notice one thing, when I was standing at a fax attched to PSTN and trying to send a fax to a fax attached to the Asterisk: The PSTN fax never switched to saying “Sending...” in the display just “Dialing”, but I can “hear” the Asterisk fax i answering. When I went back to Trunk version and did the same, I saw the fax display going from “Dialing” to “Sending” to “Sending OK”. I am sorry to say that I am not smart enough to know what trace I should start looking at, any knows? From: Vladimir Mikhelson Sent: Thursday, January 13, 2011 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: magnu...@inputinterior.se Subject: Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2 Magnus, Can it be the same as I experienced https://issues.asterisk.org/view.php?id=18542 ? Do not be confused by the ticket subject, it reflects the symptoms as they looked originally You can try the patch if applicable and if you know how to compile Addons in 1.8 separately or if you have a capacity to compile the whole thing. -Vladimir On 1/13/2011 6:31 AM, magnu...@inputinterior.se wrote: Gentlemen, We have a setup as below: PSTN – E1 – Avaya – OOH323 trunk – Asterisk – SPA-2102 – Fax machine Running Asterisk SVN-trunk-r280589M, fax working as a clock. I decided to leave “trunk” and go a stable version so I upgraded to 1.8.2. Didn’t change any config files, everything worked as before except fax. I wonder if there are any known issues or things that I have missed to do in some config file. Did a downgrade to SVN-trunk-r280589M and fax started to work again. /Magnus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-userswlEmoticon-sadsmile[1].png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday Jan 14th @ 12 Noon EST: Humbug
Greetings ${FellowVoIPuser},
When I saw the word Humbug in the Asterisk mailing list, I
remembered my friend Nir Simionovich had mentioned it to me at some
point, possibly at the big wine tasting party in Rostock during
AMOOCON, which may explain why I had forgot about it. Seeing the
thread on the ML, I checked in with Nir and Boaz and invited them to
join the VUC for a status report on the project. This should be a
great call! You can hear all about it and ask questions by joining us
live, or check later for the recorded session.
At 12 Noon EST (http://vuc.me/next for local times) dial
sip:200...@login.zipdx.com with g722 if possibly or g711. Those are
the only two codecs of ZipDX.
There is a call widget provided by PhoneFromHere.com on the home page
of the main VUC site, displayed during conference hours, as well as a
URL for the mp3 stream. Finally, you can also join by calling
skype:vuc.me
Main site for info: http://VoipUsersConference.org
Hear you there!
/r
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