[asterisk-users] Microsoft CRM Integration
Hi All, I'm hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Regards David Klaverstyn -- _ -- 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] Call Pickup how to display CND of incoming number
Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David Klaverstyn-- _ -- 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] Queues and Distinctive Ring with Alert-Info
Hi All, I'm new to Queues and I have created one as follows which seems to work ok. [david-test] strategy = rrmemory timeout = 10 retry = 0 maxlen = 0 announce-frequency = 0 announce-holdtime = no member = SIP/121 member = SIP/122 member = SIP/123 I'm wondering how do you change the SipAddHeader/Alert-Info when a call comes from a queue so users know it is a queue that is calling? Is something like the following supposed to work? exten = 0453451564,1,SipAddHeader(Alert-Info: n=Classic-4;w=3;c=4) exten = 0453451564,2,Queue(david-test) Regards David Klaverstyn -- _ -- 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] QoS : tos and cos settings
Hi All, I need some assistance with QoS. We have multiple Asterisk servers on a MPLS network. We have just moved over to Verizon and for us to get QoS Verizon are saying we need to use af41. I need to check what exactly I need to do as this is a new area for me. We only use IAX over our WAN links as SIP is on the local LAN. In IAX I have set: /etc/asterisk/iax.conf tos=af41 cos=4 https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service this pages sais I need to use a vconfig command. This is what is confusing me. vconfig set_egress_map [vlan-device] [skb-priority] [vlan-qos] is someone able to suggest what I should use here? I'm guessing my iax.conf setting is correct. Your help is greatly appreciated. Regards David-- _ -- 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] Polycom, Dial Specific Number on Handset Pickup
Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David.-- _ -- 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] BRI Installation
Hi Guys, All my installs are based on PRI ISDN. I now have a site that I need to install BRI. As I have not done a BRI install before I'm wanting to get some information from the people in the know if I need to do anything special. Typically I install libpri, dahdi Linux and tools, asterisk ...and then configure dahdi as one does for the required hardware. Is the same true for BRI with the exception of the libpri? I have this feeling that I need to install some other Linux drivers or something for BRI. I've purchased a Digium HB8 card and I don't see any mention of this in /etc/dahdi/modules. I've looked over the documentation at https://www.digium.com/en/supportcenter/documentation/viewdocs/H8 but there doesn't seem to be anything there that tells me how to configure dahdi or asterisk. If someone could give me some direction that would be greatly appreciated. Regards David-- _ -- 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] Recommendations on FXS Bank
Hi All, Can someone please recommend a FXS bank that support a minimum of 12 ports. I would prefer an IP connection to Asterisk rather than a USB or physical card. If an IP style is not available I'll consider a USB type. A card is not an option. Your recommendations would be greatly appreciated. Regards David Klaverstyn-- _ -- 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] Can run from shell but not from Asterisk System command
Hi All, I have this strange problem on a newly installed PBX. 1.8.12.0. I have other installs of 1.8.12.0 that does not exhibit this problem. I can run from the console /usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61123123 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59 ... and an email will be emailed to me. The following does not produce an email. exten = 1122,1,Answer exten = 1122,n,System(/usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61123123 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59) exten = 1122,n,HangUp result from CLI == Using SIP RTP CoS mark 5 -- Executing [1122@internal:1] Answer(SIP/8930-002b, ) in new stack -- Executing [1122@internal:2] System(SIP/8930-002b, /usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61735108900 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59) in new stack -- Executing [1122@internal:3] Hangup(SIP/8930-002b, ) in new stack == Spawn extension (internal, 1122, 3) exited non-zero on 'SIP/8930-002b' The log files that the fax2mail script generates are all correct but no email. Regards David Klaverstyn -- _ -- 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] Can run from shell but not from Asterisk System command
Sorry I meant to mention that Asterisk is running as root. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David C Sent: Friday, 11 May 2012 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can run from shell but not from Asterisk System command Hi All, I have this strange problem on a newly installed PBX. 1.8.12.0. I have other installs of 1.8.12.0 that does not exhibit this problem. I can run from the console /usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61123123 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59 ... and an email will be emailed to me. The following does not produce an email. exten = 1122,1,Answer exten = 1122,n,System(/usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61123123 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59) exten = 1122,n,HangUp result from CLI == Using SIP RTP CoS mark 5 -- Executing [1122@internal:1] Answer(SIP/8930-002b, ) in new stack -- Executing [1122@internal:2] System(SIP/8930-002b, /usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61735108900 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59) in new stack -- Executing [1122@internal:3] Hangup(SIP/8930-002b, ) in new stack == Spawn extension (internal, 1122, 3) exited non-zero on 'SIP/8930-002b' The log files that the fax2mail script generates are all correct but no email. Regards David Klaverstyn -- _ -- 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] Script to automatically update externip. Useful for a host with dynamic public IP
To me it would be simpler to use externhost instead of externip and then use a dynamic DNS service. It has worked flawlessly for me for many years. Regards David. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Raj Mathur (??? ?) Sent: Tuesday, 7 February 2012 1:19 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Script to automatically update externip. Useful for a host with dynamic public IP On Monday 06 Feb 2012, John Cahill wrote: logger -s checksetexternip.sh: External IP address has changed, changing /etc/asterisk/sip_general_custom.conf grep -v externip /etc/asterisk/sip_general_custom.conf /etc/asterisk/sip_general_custom.conf.tmp echo externip=$EXTERNIP /etc/asterisk/sip_general_custom.conf.tmp cp /etc/asterisk/sip_general_custom.conf.tmp /etc/asterisk/sip_general_custom.conf rm /etc/asterisk/sip_general_custom.conf.tmp You could also do something like: sed -i -e s/^externip *=.*/externip = $EXTERNIP/ /etc/asterisk/sip.conf Apologies for the wrapped code. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Anyone have a reliable T.38 Solution
I'm using 1.8, but also have 1.4 and 1.2 installs using the same configuration. Regards David. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell Sent: Thursday, 5 January 2012 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anyone have a reliable T.38 Solution On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn da...@klaverstyn.com.au wrote: I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and rx_fax on multiple installations with no problems. David, Are you running 10.0 or 1.8? Glad to know that the PAP2T has a solid T.38 implementation! -Matt -- _ -- 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] dahdi_tool missing
Hi All, I have installed newt and newt_devel but dahdi_tool will not compile/install. I'm trying this with dahdi-linux-complete-2.5.0.2+2.5.0.2. does anyone have any suggestions as to what I am doing wrong? Regards David.-- _ -- 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] dahdi_tool missing
There is no error message. All the other tools are built excepts for dahdi_tool. Regards David. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, 22 December 2011 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi_tool missing On Thu, Dec 22, 2011 at 12:14:55AM +, Klaverstyn, David C wrote: I have installed newt and newt_devel but dahdi_tool will not compile/install. I'm trying this with dahdi-linux-complete-2.5.0.2+2.5.0.2. does anyone have any suggestions as to what I am doing wrong? What is the specific error message you're getting? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 -- _ -- 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] Split E1 ISDN service for another device.
Hi All, I know this is not specifically Asterisk related but I don't knew where else to ask for help. Does anyone know how to or if it is even possible to allocate 512kbit/s to an ISDN device from a 30B+D ISDN line. The building the office is in has a E1 30 channel service (30B+D) but we can not get any 2B+D ISDN services. I have a HDX Polycom video conferencing system that requires a 512kbit/s service. I am told to allocate 8B+D service from the 30B+D to the Polycom device.Is this even possible. I have a Digium TE121 currently install in the server that the E1 ISDN line is connected to. The Polycom has 4 by RJ45 connections for the 512kbit/s service. Any help would be appreciated. Regards David. -- _ -- 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] dahdi not available in Asterisk
Hi All, I must be doing something really stupid as I can't get DAHDi working in Asterisk. It is loaded and working in Linux fine. *CLI module load chan_dahdi Unable to load module chan_dahdi Command 'module load chan_dahdi' failed. [2010-03-08 14:05:34 CST] WARNING[26926]: loader.c:393 load_dynamic_module: Error loading module 'chan_dahdi': libpri.so.1.4: cannot open shared object file: No such file or directory [2010-03-08 14:05:34 CST] WARNING[26926]: loader.c:770 load_resource: Module 'chan_dahdi' could not be loaded. From the Log file. [2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Error loading module 'chan_dahdi.so': libpri.so.1.4: cannot open shared object file: No such file or directory [2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Module 'chan_dahdi.so' could not be loaded. I am using on CentOS 5.4 64 bit. Asterisk1.6.0.25 Asterisk-addons 1.6.0.4 Libpri 1.4.10.2 I have install libpri first and then asterisk. Regards David. -- _ -- 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] SIP client MAC address.
From Linux you could use arp | grep 192.168.0.1 substituting the IP address of the SIP device. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client MAC address.
If there is more than one SIP devices operating from the same NAT device then I'm not sure what you could do as it would always show the same IP for all SIP devices behind the same NAT. If there is only one device behind that NAT making a connection to your server then that is easy, if not I think your screwed. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP client MAC address. hello david, what in case of sip client is behind NAT, and i want SIP client IP address. not from system from which client registered. if it is a SIP phone then what? if you have any idea then tell me. regards dhaval On Wed, Oct 28, 2009 at 12:02 PM, Klaverstyn, David C david.klavers...@intergraph.com wrote: From Linux you could use arp | grep 192.168.0.1 substituting the IP address of the SIP device. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. hello, is there any facility to get SIP client (ex. softphone,ipphone) MAC address on asterisk. based on that we authenticated client in anyway. i tried with sip debug but i didn't got any MAC address related field in all packets. regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Call rejected, CallToken Support required
Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the calltokenignore list or setting user priv requirecalltoken=no I have seen posts that suggest using: calltokenoptional = 0.0.0.0/0.0.0.0 or calltokenignore=xxx.xxx.xxx.xxx Using the above cause asterisk not to display the error but nothing occurs in the CLI. If I enable debug I see the following with the option calltokenoptional = 0.0.0.0/0.0.0.0 in iax2.conf in the general section. On the sending Server Asterisk 1.2.x Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 7ms SCall: 01471 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 138954087 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 01471 [192.168.42.251:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00010ms SCall: 03923 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 182789945 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 03923 [192.168.42.251:4569] Tx-Frame Retry[000] -- OSeqno: 091 ISeqno: 076 Type: VOICE Subclass: 136 Timestamp: 1048584ms SCall: 1 DCall: 2 [192.168.22.251:4569] Oct 2 10:05:41 NOTICE[32273]: chan_iax2.c:2880 auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.42.250:4569-4 is circuit-busy Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP On the receiving Server Asteirsk 1.4.x Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 00657 DCall: 2 [192.168.25.250:4569] AUTHMETHODS : 3 CHALLENGE : 152361611 USERNAME: priv Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 2 DCall: 00657 [192.168.25.250:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04012ms SCall: 2 DCall: 0 [192.168.25.250:4569] CAUSE CODE : 0 I would really appreciate it if someone was able to give me an answer to this problem or at least point me in the right direction. Regards David. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 Call rejected, CallToken Support required
I tried your recommendation. I don't get an error with that but the call is cancelled with a debug of: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00011ms SCall: 00269 DCall: 3 [192.168.25.250:4569] AUTHMETHODS : 3 CHALLENGE : 182763616 USERNAME: priv Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 3 DCall: 00269 [192.168.25.250:4569] Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04011ms SCall: 3 DCall: 0 [192.168.25.250:4569] CAUSE CODE : 0 -Original Message- From: cov...@ccs.covici.com [mailto:cov...@ccs.covici.com] Sent: Friday, 2 October 2009 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Klaverstyn, David C Subject: Re: [asterisk-users] IAX2 Call rejected, CallToken Support required I had a problem between my 1.6.0 server and a 1.4 server trying to call through iax and I just put requirecalltoken=no in the stanza and that fixed the problem. Klaverstyn, David C david.klavers...@intergraph.com wrote: Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the calltokenignore list or setting user priv requirecalltoken=no I have seen posts that suggest using: calltokenoptional = 0.0.0.0/0.0.0.0 or calltokenignore=xxx.xxx.xxx.xxx Using the above cause asterisk not to display the error but nothing occurs in the CLI. If I enable debug I see the following with the option calltokenoptional = 0.0.0.0/0.0.0.0 in iax2.conf in the general section. On the sending Server Asterisk 1.2.x Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 7ms SCall: 01471 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 138954087 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 01471 [192.168.42.251:4569] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00010ms SCall: 03923 DCall: 4 [192.168.42.251:4569] AUTHMETHODS : 3 CHALLENGE : 182789945 USERNAME: priv Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 03923 [192.168.42.251:4569] Tx-Frame Retry[000] -- OSeqno: 091 ISeqno: 076 Type: VOICE Subclass: 136 Timestamp: 1048584ms SCall: 1 DCall: 2 [192.168.22.251:4569] Oct 2 10:05:41 NOTICE[32273]: chan_iax2.c:2880 auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.42.250:4569-4 is circuit-busy Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP On the receiving Server Asteirsk 1.4.x Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 00657 DCall: 2 [192.168.25.250:4569] AUTHMETHODS : 3 CHALLENGE : 152361611 USERNAME: priv Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 2 DCall: 00657 [192.168.25.250:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04012ms SCall: 2 DCall: 0 [192.168.25.250:4569] CAUSE CODE : 0 I would really appreciate it if someone was able to give me an answer to this problem or at least point me in the right direction. Regards David. Alternatives: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
[asterisk-users] TE121P Blue Alarm/Recovering
Hi All, I have a TE121P card installed and since connected it to the PRI I keep getting the Current Alarm as continually changing from Blue Alarm/Recovering and Recovering. The config I have is: /etc/dahdi/system.conf bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone= cn defaultzone = cn /etc/asterisk/chan_dahdi.cfg [channels] Context=telco language=cn switchtype=euroisdn signalling=pri_cpe rxwink=300 usecallerid=yes . . echocancel=yes group=1 channel=1-15,17-31 I installed the following components Asterisk 1.4.26.2 DAHDI Linux 2.2.0.2 DAHDI Tools 2.2.0 Libpri 1.4.10.1 Addons 1.4.9 Any help would be greatly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE121P Blue Alarm/Recovering
I'd appreciate it if someone was able to assist. Running the command: dahdi_hardware: pci::08:08.0 wcte12xp+d161:8000 Wildcard TE121 dahdi_scan: [1] active=yes alarms=REC description=Wildcard TE121 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE121 with VPMADT032 location=PCI Bus 08 Slot 09 basechan=1 totchans=31 irq=169 type=digital-E1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS cat /etc/dahdi/system.conf: bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn From: Klaverstyn, David C Sent: Monday, 28 September 2009 5:05 PM To: 'asterisk-users@lists.digium.com' Subject: TE121P Blue Alarm/Recovering Hi All, I have a TE121P card installed and since connected it to the PRI I keep getting the Current Alarm as continually changing from Blue Alarm/Recovering and Recovering. The config I have is: /etc/dahdi/system.conf bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone= cn defaultzone = cn /etc/asterisk/chan_dahdi.cfg [channels] Context=telco language=cn switchtype=euroisdn signalling=pri_cpe rxwink=300 usecallerid=yes . . echocancel=yes group=1 channel=1-15,17-31 I installed the following components Asterisk 1.4.26.2 DAHDI Linux 2.2.0.2 DAHDI Tools 2.2.0 Libpri 1.4.10.1 Addons 1.4.9 Any help would be greatly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE121P Blue Alarm/Recovering
Many thanks John of Sydney. I removed the CRC4 and it worked straight away. Can you recommend a CRC type at all or would it be best to leave it as nothing? David of Brisbane. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee, John (Sydney) Sent: Tuesday, 29 September 2009 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE121P Blue Alarm/Recovering 1) I have not seen a blue light (usually red/yellow) before on a Digium card and so don't really know what it means. 2) Try to see if you can see any messages coming up from the Asterisk box itself (not thru putty or other remote console). You should see a steady stream of error messages coming up. Also, look at /var/log/asterisk/messages or event_log. There may or may not be anything there. 3) I see that loadzone = cn which means the installation is in China. 4) My experience tells me that if it is in major China cities, the ISDN line would be E1 which is correct but from the installations I did, China's E1 does not like CRC4. 5) I think it is highly likely to be a ISDN line config (telco or Asterisk side) problem. 6) You can try contacting Digium support too. They provide prompt support but you have to register you serial no first. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David C Sent: Tuesday, 29 September 2009 10:07 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TE121P Blue Alarm/Recovering I'd appreciate it if someone was able to assist. Running the command: dahdi_hardware: pci::08:08.0 wcte12xp+ d161:8000 Wildcard TE121 dahdi_scan: [1] active=yes alarms=REC description=Wildcard TE121 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE121 with VPMADT032 location=PCI Bus 08 Slot 09 basechan=1 totchans=31 irq=169 type=digital-E1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS cat /etc/dahdi/system.conf: bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn From: Klaverstyn, David C Sent: Monday, 28 September 2009 5:05 PM To: 'asterisk-users@lists.digium.com' Subject: TE121P Blue Alarm/Recovering Hi All, I have a TE121P card installed and since connected it to the PRI I keep getting the Current Alarm as continually changing from Blue Alarm/Recovering and Recovering. The config I have is: /etc/dahdi/system.conf bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn /etc/asterisk/chan_dahdi.cfg [channels] Context=telco language=cn switchtype=euroisdn signalling=pri_cpe rxwink=300 usecallerid=yes . . echocancel=yes group=1 channel=1-15,17-31 I installed the following components Asterisk 1.4.26.2 DAHDI Linux 2.2.0.2 DAHDI Tools 2.2.0 Libpri 1.4.10.1 Addons 1.4.9 Any help would be greatly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement
Faxing over SIP never worked for me. The faxes would always fail. When I saw the information about T.38, I decided to immediately upgrade to 1.6.0.11-rc2 from 1.6.0.10 and try it. I was amazed. Without having to change anything in my configuration faxes just worked. I have tested it with multiple faxes, short and long, and faxes with images and they all came through. Well done guys. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Team Sent: Monday, 3 August 2009 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1,and 1.6.2.0-beta4 Release Announcement The Asterisk Development Team is pleased to announce the the second release candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of 1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ . The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The related security advisory AST-2009-004 has been released along with this announcement. Please read that advisory for more information. The release candidates and betas, in addition to other fixes, contain a major re-work of the T.38 support in Asterisk. If you've been having trouble with T.38 in the 1.6 series, you are strongly encouraged to try one of these release candidates to determine if these changes fixed your T.38 issues. For a full list of changes in these releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.0.11-rc2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.1.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.1.3-rc1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.2.0-beta4 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building a System.
Hi John, I'm not sure if this will help you or not but I created a script that will install Asterisk with all the required components for DAHDI, Faxing, fax to email, LDAPget, CDR, FOP etc. It can even include text to speech applications. I created it because I wanted to install Asterisk multiple times and as quickly as possible. It does the exact same steps as one would do when doing an install manually. I created the bash script aster-install http://www.klaverstyn.com.au/wiki/index.php?title=Aster-install . http://www.klaverstyn.com.au/wiki/index.php?title=Aster-install Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John F. Ervin Sent: Monday, 11 May 2009 1:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Building a System. So, people have recommended building a system from scratch, start with a CentOS base and installing asterisk and all of the other utilities. I've only used Trixbox for my business system. I'm wondering what surprises I'd run into. Right now, I know I'd need the OS, Asterisk, something like FreePBX, I have a x100p card so I'd need Zaptel, does that come with asterisk? Fax support, seems to work with Trixbox, but I've heard that it needs to be loaded. Voicemail etc.? I mean, I don't know exactly what you'd need because almost everything I need comes with the Trixbox build. Are there (??) instructions for people who are experienced at the Trixbox level but wish to move on? -- John F. Ervin Central Florida TeleSource, LLC. 4270 Aloma Ave #124-69C Winter Park, FL 32792 (W) 407-679-6238 (F) 866-566-1282 (F) 321-445-0781 jer...@jervin.com http://jervin.com/cft ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 64bit: any problems with asterisk?
Is there something that you need to do so Asterisk will compile to 64bit or will Asterisk just function as 32bit on a 64bit platform? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Sunday, 26 April 2009 2:43 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 64bit: any problems with asterisk? John Novack wrote: Suggest you use CentOS rather than Fedora. CentOS has a longer support life, with the same cost. JMO John Novack sean darcy wrote: We're getting a new server. I'm considering installing 64bit fedora rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any issues we should expect? sean Thanks for all the responses. I didn't expect any issues with 64 bit, but... So I'm off to install this weekend. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Records are not working
I have the same problem for Disposition when I use call files. The
duration is correct but the Disposition is always NO ANSWER. I also am
using 1.6.0.1. I did not have the problem when I was using 1.4.21
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedram M
Sent: Monday, 27 October 2008 10:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR Records are not working
Hello Asterisk-Users,
For some reason my CDR records for disposition and billsec are not
working correctly.
I always receive a 0 for billsec and the disposition is always at NO
ANSWER', even when I grab the calls.
I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22.
Here is information on how I do the call:
-
.call file contents:
-
Channel: SIP/GAFACHI/1818345
CallerID: 18183455512
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Context: outboundmessage1
Extension: s
Priority: 1
Set: PassedInfo=18183453041-m1d
-
extensions.conf for outboundmessage1 context:
-
[outboundmessage1]
exten = s,1,Set(CDR(userfield)=${PassedInfo})
exten = s,2,Answer
exten = s,3,System(/opt/asterisk/scripts/custom/answer.sh ${CDR(clid)}
${CDR(dst)} ${CDR(billsec)})
exten = s,4,Background(/tmp/hello-world)
exten = s,5,WaitExten()
-
Master.csv after picking up the line:
-
,1818345512,2,outboundmessage1,1818345,SIP/GAFACHI-090cd7
90,,Hangup,,2008-10-27 00:19:08,,2008-10-27 00:19:33,25,0,NO
ANSWER,DOCUMENTATION,1225066748.1,18183453041-m1d
Any insight would be appreciated.
Thanks,
Pedram
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[asterisk-users] Asterisk CDR Analyser
Hi All,
I'm stuck and need some help. I have installed the Asterisk CDR
Analyser Version 2.0.1. It mostly works except for the CDR Report. I
get the following error even though it lists the CDR details.
Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day,
sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE
UNIX_TIMESTAMP(calldate) = UNIX_TIMESTAMP('2008-10-01') GROUP BY
substring(calldate,1,10)
MySQL Error: 1146 (Table 'asterisk.cdr' doesn't exist)
From memory there is a file that needs to be modified that is hard coded
to use the table asterisk.cdr but I can't find it anywhere.
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Re: [asterisk-users] Asterisk CDR Analyser
Brilliant, many thanks. It is now working once I changed it to the
correct table name. Line 232 is correct as well, column 103
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Bright
Sent: Friday, 10 October 2008 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk CDR Analyser
That query appears in call-log.php around line 232.
On Thu, Oct 9, 2008 at 10:20 PM, Klaverstyn, David C
[EMAIL PROTECTED] wrote:
Hi All,
I'm stuck and need some help. I have installed the Asterisk CDR
Analyser
Version 2.0.1. It mostly works except for the CDR Report. I get the
following error even though it lists the CDR details.
Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day,
sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE
UNIX_TIMESTAMP(calldate) = UNIX_TIMESTAMP('2008-10-01') GROUP BY
substring(calldate,1,10)
MySQL Error: 1146 (Table 'asterisk.cdr' doesn't exist)
From memory there is a file that needs to be modified that is hard
coded to
use the table asterisk.cdr but I can't find it anywhere.
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[asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0
Mysql for CentOS 5.2 is the mysql client tools. mysql.i386 : MySQL client programs and shared libraries. Does anyone have any other suggestions? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: Tuesday, 7 October 2008 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't find mysqlclient : asterisk-addons-1.6.0 Yes, unfortunately, VOIP wiki did not mention about installing mysql-client which it should have been. Without mysql-client, you cannot change passwords, grants, etc. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stefan Schmidt Sent: Tuesday, 7 October 2008 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't find mysqlclient : asterisk-addons- 1.6.0 Klaverstyn, David C schrieb: Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. Hello, maybe you should install mysql-client too ;) best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming MoH on 1.4
You may need to wait up to 45 seconds after asterisk first starts to buffer a bit of the stream. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Monday, 15 September 2008 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Streaming MoH on 1.4 Nope, doesn't seem to work. all I hear is deadair. Original Message Subject: Re: [asterisk-users] Streaming MoH on 1.4 From: Mark Hamilton [EMAIL PROTECTED] Date: Sun, September 14, 2008 8:48 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Let me test it out.. Original Message Subject: Re: [asterisk-users] Streaming MoH on 1.4 From: Klaverstyn, David C [EMAIL PROTECTED] Date: Sun, September 14, 2008 8:37 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com This works for me. [SkyFM-80s] mode=custom application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://160.79.128.62:6618 http://160.79.128.62:6618 [SkyFM-HotHits] mode=custom application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://160.79.128.62:6628/ http://160.79.128.62:6628/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Mark Hamilton Sent: Monday, 15 September 2008 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Streaming MoH on 1.4 I'm still searching, but can't find anything anywhere other than the part that just doesn't work which I've tried already. Original Message Subject: [asterisk-users] Streaming MoH on 1.4 From: Mark Hamilton [EMAIL PROTECTED] Date: Sun, September 14, 2008 12:56 am To: Asterisk Mailing asterisk-users@lists.digium.com Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92 http://nerdvittles.com/index.php?p=92 ) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream = quietmp3:/var/lib/asterisk/mohmp3/stream,http://wbez-sclo.streamguys.us/ http://wbez-sclo.streamguys.us/ ;application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://wbez-sclo.streamguys.us/ http://wbez-sclo.streamguys.us/ Every write up about this has done either the above, or a variant of the above (like the commented out application line) Nothing works. Using the recommended old version of mpg123 0.59r (which gives that stupid monmp3thread: request to schedule in the past error) When I say it doesn't work, I dial the extension, CLI says started musiconhold, and immediately followed by stopped musiconhold. Sometimes between these two messages there's the request to schedule in the past thing. I know the error/warning is due to ztdummy timing (which I currently don't have right now) but it's still not the big problem. Can somebody please help me out with this? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF call pickup on Linksys SPA932
When I try this my GrandStream GXP-2000 gives me an error 603, which is
Declined.
-- Executing Pickup(SIP/8908-b7987738, SIP/[EMAIL PROTECTED]) in new
stack
== Spawn extension (internal, **8948, 1) exited non-zero on
'SIP/8908-b7987738'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Friday, 12 September 2008 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] BLF call pickup on Linksys SPA932
From memory, this is an issue with Asterisk 1.2 which can be fixed by
moving to 1.4
PaulH
Chris Bagnall wrote:
Greetings list,
We recently installed some Linksys SPA962 + SPA932 sidecars into a
client's offices. The BLF functionality works fine, but call pickup
using the BLF is something of an issue.
Following the advice on voip-info.org, I configured part of their
dialplan as follows:
exten = _**2XX,1,Pickup(SIP/${EXTEN:2})
exten = _**2XX,n,Dial(SIP/${EXTEN:2},15,tw)
exten = _**2XX,n,Voicemail(${EXTEN:[EMAIL PROTECTED],su)
The BLFs are all registered in the form:
exten = **201,hint,SIP/201
etc.
The theory being that pressing the BLF for 201, for example, will try
and pick up any ringing call on SIP/201, then, if that fails (i.e.
there's no call to pick up), the extension will then be dialled as
normal.
Problem is, Pickup() seems to die if there's no call to pick up,
rather than continue dialplan execution.
Does anyone know a way round this, or a better way of doing it?
Thanks in advance.
Regards,
Chris
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[asterisk-users] Digium Hardware Echo Cancellation
Hi All, When using a Digium card with a hardware echo cancellation module installed, is the only thing required to enable it, is to set echocancel=yes in zapata.conf? I think I remember seeing somewhere that you need to make a change in the zaptel Makefile to allow the echo cancellation to work. Is that correct and if so what is it? Regards David. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sharing unused minutes between Asterisk users
Hi All, I was under the impression that I found a WEB site about two years or so ago that allowed Asterisk users to place free calls between each other that used up users un-used minutes/calls. I though the site was IAXtel but that does not seem to be the case. As an example I have a plan with a VSP. They allow a certain number of calls every month but I only use 20% of the allocation. I was wanting to let other people around the world to utilise the additional calls I have. Is there such a WEB site that allows us to connect our Asterisk servers together to utilise the otherwise unused calls? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Loosing SIP registration.
Hi All, I am having problems with some SIP peers. I seem to loose registration. If I reload SIP the registration comes back. They usually stay registered for about 2 days before they drop. The problem is not all of them drop usually just the list 2 in the list. The other strange thing is that the 2 the do drop their registration do not occur at the exact same time. It could be many hours between them. I am using Asterisk 1.4.18.1 Any help would be greatly appreciated. My parent's server is having the problems. My server does not exhibit this problem. I just took my router/firewall down to them as I have just purchased a new one and they are still experiencing the problem. sip show registry Host Username Refresh StateReg.Time 202.168.56.133:506061990xx 105 Registered Fri, 11 Apr 2008 15:15:58 sip.pennytel.com:5060 61289xx 105 Request Sent Thu, 10 Apr 2008 21:38:54 sip2.bbpglobal.com:5060 617000xxx 105 Request Sent Thu, 10 Apr 2008 20:43:20 sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip-register.conf': Found == Parsing '/etc/asterisk/sip-klavo.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found sip show registry Host Username Refresh StateReg.Time 202.168.56.133:506061990xx 105 Registered Fri, 11 Apr 2008 15:16:15 sip.pennytel.com:5060 61289xx 105 Registered Fri, 11 Apr 2008 15:16:16 sip2.bbpglobal.com:5060 617000xxx 105 Registered Fri, 11 Apr 2008 15:16:16 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Touch Recording
Thanks for that. I have the timeout set to 3000 ms and I have been pressing the *1 within 500 ms so I don't think it is related to that. As I can do it over SIP but not ZAP does not make much sense to me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: Monday, 7 April 2008 3:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One Touch Recording I had this problem before...the following appeared in a previous post... For some reasons, the * and 1 must be pressed pretty quickly together on the Polycom phone before it can be transmitted successfully to Asterisk. Does anyone know if that can be tuned? Sure... go to features.conf, and change the value of the featuredigittimeout option. Hope this helps. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Monday, 7 April 2008 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] One Touch Recording Hi All, For some reason One Touch Recoding does not work over ZAP but it does work when I call another extension. Both Dial commands have the W option for the calling party to enable recording. Does anyone know why it works internally but not over ZAP. I have a TE110P card on an E1 connection. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One Touch Recording
Hi All, For some reason One Touch Recoding does not work over ZAP but it does work when I call another extension. Both Dial commands have the W option for the calling party to enable recording. Does anyone know why it works internally but not over ZAP. I have a TE110P card on an E1 connection. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP650 console with expansion modules
http://www.testforme.com/download/polycom/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Thursday, 13 March 2008 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IP650 console with expansion modules Hi Bill - I just replaced an IP 601 with a new IP 650. We have 2 expansion modules attached. The lights on the expansion modules light up if a users gets an INBOUND DID call, but the lights don't light up if the user makes an OUTBOUND call. Sip: 2.1.1.0052 Has anyone seen this? Haven't seen that, but you might consider upgrading to 2.1.2 or 2.2.2. I checked the release notes and there are a few bug fixes related to BLF features. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with rxfax
Hi All, I have installed agx-ast-addons-1.4.5 on Asterisk version 1.4.18. The problem I have is that RxFAX will not answer an incoming fax. When you call the number there is just silence. This is over SIP and not ZAP. The modules rx and tx fax seem to be loaded OK. core show applications like fax -= Matching Asterisk Applications =- NVFaxDetect: Detects fax sounds on all channel types (IAX and SIP too) RxFAX: Receive a FAX to a file TxFAX: Send a FAX file -= 3 Applications Matching =- Can anyone shed some light onto this problem? -- Executing [EMAIL PROTECTED]:1] Set(SIP/09123595-08a389c0, FAXFILE=/var/spool/asterisk/fax/_1204000167.4) in new stack -- Executing [EMAIL PROTECTED]:2] Answer(SIP/09123595-08a389c0, ) in new stack -- Executing [EMAIL PROTECTED]:3] RxFAX(SIP/09123595-08a389c0, /var/spool/asterisk/fax/_1204000167.4.tif|caller) in new stack ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple Asterisk Servers. One Conference
Hi guys, I currently have about 10 Asterisk servers scattered around the place each hosting their own dynamic conference centre. Is there any way that when people join these conference centres on each server that somehow Asterisk bridges the conference centres on each server to form one large conference? Many Thanks David. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attendant phone
To me it sounds like you should be using the Flash Operator Panel to monitor that many extensions. The Polycom 6xx range can monitor 42 extensions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis Sent: Thursday, 14 February 2008 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Attendant phone As far as I'm aware, only the Aastra 57i with three 560M modules would come close to your requirements. The 57i can display up to 5 extensions at one time with a further 15 being available by the use of multiple pages. The 560M modules can display up to 20 extensions at one time with three pages being available for a total of 60 extensions per phone. This gives you a total of 200 extensions that can be monitored. voip crazy wrote: Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones are? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 320 Issue
Hi All, I'm not sure if this is related directly to asterisk or not but on my Polycom 320 when I try to dial a number smaller than 4 digits I get an error on the phone saying Enter more digits. The dial plan section is listed below. dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=x.|*x. dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address= dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=000 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan Any help would be appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 320 Issue
Sorry everyone. There was an error in the dial plan in Asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Monday, 21 January 2008 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom 320 Issue Hi All, I'm not sure if this is related directly to asterisk or not but on my Polycom 320 when I try to dial a number smaller than 4 digits I get an error on the phone saying Enter more digits. The dial plan section is listed below. dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=x.|*x. dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address= dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=000 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan Any help would be appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_rxfax.c and app_txxfax.c where?
Hi All, Where can I find copies of the app_rxfax.c, app_txfax.c and apps_Makefile.patch. They don't seem to be located at soft-switch.org anymore. I am currently trying to compile Asterisk 1.2.26.1 and need the fax components. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Traditional' Faxing
I am also very interested in these scripts. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, 13 November 2007 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 'Traditional' Faxing - Original Message - From: Jonn R Taylor [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, November 10, 2007 5:45 PM Subject: Re: [asterisk-users] 'Traditional' Faxing Greg Cockburn wrote: Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there maybe someway to bridge an Analogue FXS port to a channel on the E1? Basically I want to mimic what we have now. 1. Any person can send a fax using the fax machine, and the PBX picks the next free channel on the E1. 2. A fax call can come over any channel on the E1, and the dialed number is matched and sent to the analogue FXS port of the PBX to be received by the fax machine. Is there anyway I can do this in Asterisk that will work seamlessly? I have not yet purchased any hardware, so recommendations would be greatly appreciated. (I believe some of the problem exists due to timing, does any hardware; E1 card / Analogue card; support linking a timing signal together?) Sangoma, Digium, Pika? Thanks all for any help on this one. Greg. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Greg, There are alot of option for handeling faxes. One is to use iaxmodem and hylafax. This option works the best. You can try to use an analog adapter or card to connect a conventional fax to but this is not allways reliable. I have spent alot of time working on faxing with asterisk. If you need any help you can email me and I will send the links and scripts that I have to help you in your setup. FYI, They are all for RH/CentOS. Hardware, how many phone and trunks do you plan on using? Digium cards for analog phone's and faxes work very well, linksys makes very good ATA's too. Digium or Sangoma T1 cards are the most suppoted that I have seen. but there are others. OS, there are alot of different *nix OS's that are out there. Pick the one that you are the most comfortable to use. Asterisk was developed on RedHat though. Depending on your needs for support I would suggest either EL4 or CentOS4 with Asterisk 1.2. There are alot of people running 1.4 in production but the commercaial version of Asterisk is still on 1.2 John, Do you mind posting a link to those CentOS scripts here ? Thanks. Dovid ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
I've had experience with Linksys and Polycom. Either one is easy enough to provision. Took me a while to understand how to provision Polycom. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 30 October 2007 3:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get TCP access to CDR Master.csv
I’m no expert in this field bit I would have though logging the calls to MySQL and then queering the MySQL database would be the best not to mention the easiest way to get the details you are looking for. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Hass Sent: Thursday, 25 October 2007 8:39 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to get TCP access to CDR Master.csv Hello, I am not sure if I totally understand the question but if your looking to stream the connection you could create a simple bash script like this #!/bin/bash while true; do tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l done There probably is a better solution then this, but this will get you going From any machine you should be able to type `telnet ip.of.machine 1024` --John [EMAIL PROTECTED] wrote: Hi. I‘d like to get access to the CDR‘s generated by Asterisk (1.4) in real-time from a remote connection coming in on TCP. Basically what I have is a Windows application that is used to process incoming, outgoing and missed call records putting them into a database for some analysing etc. This app can connect to a TCP server and read from this connection the CDR‘s as they are coming in (being generated). I can‘t find this as a „feature“ of the standard Asterisk... but maybe I‘m missing something? The closest I could get is something around the manager api but it‘s not really what I‘m after. I‘d like to access the CDR‘s them selves. Being a (more or less) novice Linux user the only thing I can think of is trying to do this using Perl scripts where it would set up a listening socket and when connection is received it would do something like (in princip, not managed to do this properly yet): ... print $connection `tail –f /var/log/asterisk/cdr-custom/Master.csv` ... But even this is full of issues to solve. Things like only one connection at a time (which I can live with) from the remote computer. The fact that tail will not write to the socket (yeah, a major issue probably) which I‘m thinking of trying to solve by reading line by line somehow and writing back to the socket... not even sure if this is possible. So basically I‘m hoping someone has a nice solution for this. With or witout scripting, external programs of some sort (runnin ubuntu 7.04 or 6.06) or whatever works. I‘d really appreciate your input here. Sincerely, Baldvin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dedicated Codec Conversion Server
The way I would accomplish this is to have 2 Asterisk boxes. Your
conversion server would just have a dial plan to forward all calls to
the Asterisk box that has the PSTN interface. Once the PSTN Asterisk
Server receives the calls it just routes the call based on dial plan
rules.
{Internet - VPN} - Conversion Server - Asterisk PSTN.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, 25 October 2007 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dedicated Codec Conversion Server
I have a need for a large number of remote phones. I want to use GSM
between the phones and the conversion server which will transcode to
ulaw eventually send or recieve calls via the PSTN (ulaw).
I am curious is anyone has any ideas on the easiest way to create a
dedicated codec conversion box. It will be running openvpn and so will
the remote PCs with softphones (x-lite).
So I want the remote softphones to connect to the codec conversion
asterisk box and then send the call to the main Asterisk server as ulaw
and pass call in and out the pstn as ulaw.
Any ideas for a simple implementation without creating all kinds of
funky conf files. Seems simple but the solution eludes me (maybe
because I have been working over 18 hours.
Thanks,
Steve Totaro
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Re: [asterisk-users] AA50 Paging
I’m not sure if this will work on the Grandstream phones but I use this for the Linksys phones. exten = ,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = ,n,Dial(SIP/201) exten = ,n,HangUp I would guess it would work with multiple phones, i.e., exten = ,n,Dial(SIP/201 SIP/202 SIP/203 SIP/204) You may need to check the phone is configured for paging auto answer. The Linksys has a field of Paging Serv and is set to yes. Let me know if it works. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal Sent: Monday, 15 October 2007 7:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AA50 Paging Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell PowerEdge 860, Sangoma A108
I use the PE 860 servers in our installs. I even have some in a cluster for the remote sites we support. I have only used Digium TE1x0P card in these servers with no problems at all. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Girts Graudins Sent: Tuesday, 9 October 2007 8:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dell PowerEdge 860, Sangoma A108 Hello everyone, I'm considering getting me a quad-core Dell PowerEdge 860 to run Asterisk. Anyone else using this model? Any tales of woe and sorrow I should know about? Then, in a couple of weeks, I'm thinking of getting a Sangoma A108 and giving that a try. Same question with that one - any quirks I should be aware of? Girts ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
I've installed them in a number of sites. The phones are good and easy to provision. If you need a good speakerphone then choose another phone. If there is something specific you need to know let me know. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb Sent: Monday, 24 September 2007 3:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Anyone use the Linksys phones? Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. I have read through the literature, but would like some real world feedback. Thanks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voxalot User and Peer details.
Can someone please post a peer and user context for Voxalot. I can't seem to get one working correctly. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VSP authentication to incorrect context
All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch, have GoTalk, digest has 09xx ;GoTalk Outbound [GoTalk] username=09xx fromuser=09xx fromdomain=sip.gotalk.com type=peer secret= qualify=yes host=sip.gotalk.com disallow=all allow=g729 ;GoTalk Inbound [09xx] username=09xx type=user secret= fromuser=09xx host=sip.gotalk.com context=from-vsp canredirect=no Registration string is register=09xx:[EMAIL PROTECTED]/09xx David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] , www.intergraph.com.au http://www.intergraph.com.au ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VSP authentication to incorrect context
Many thanks, that did the trick. I actually read that page previously. I'm Not sure why it did not work or why I did not try entering that line previously. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Tuesday, 4 September 2007 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VSP authentication to incorrect context This links seems to show that insecure=very might need to be set later, PaulH http://forums.whirlpool.net.au/forum-replies-archive.cfm/359239.html On Tue, 2007-09-04 at 13:50 +1000, Klaverstyn, David C wrote: All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch, have GoTalk, digest has 09xx ;GoTalk Outbound [GoTalk] username=09xx fromuser=09xx fromdomain=sip.gotalk.com type=peer secret= qualify=yes host=sip.gotalk.com disallow=all allow=g729 ;GoTalk Inbound [09xx] username=09xx type=user secret= fromuser=09xx host=sip.gotalk.com context=from-vsp canredirect=no Registration string is register=09xx:[EMAIL PROTECTED]/09xx David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 [EMAIL PROTECTED], www.intergraph.com.au ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware download
http://www.testforme.com/download/ I'll leave the files there for a few days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Monday, 27 August 2007 4:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom firmware download Hi: Doug wrote: At 13:29 8/25/2007, Al lists wrote: Thats just sad, I got SIP 2.2 from trixbox now, but still we need to have some sort of place at least for ourselves to download this stuff. Looking for boot loader now. Which version? http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip 330_320.html#download http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip 430.html#download It's funny how every time this question gets asked, there's some smart guy (who doesn't use Polycom sets himself) who finds these links. (I'm sincerely thankful for the effort, though.) Only authorized resellers can download the current firmware from those URLs. The only guaranteed way to get the current firmware is to get it from a/your reseller. Posting the firmware packages on a third-party site is a violation of Polycom's EULA. Why do they do this? Because they want to control the sales channel. I don't agree with it, but it's how they operate. If you want a more detailed answer, ask Polycom directly, and I wish you luck. Cheers, -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom and NAT
I have both of those command lines for my natted sip device. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Wednesday, 22 August 2007 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and NAT In your sip.conf, for the user: nat=yes To send keepalives for the UDP connection (depending on how flimsy the device handling NAT is): qualify=yes From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Tuesday, August 21, 2007 17:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom and NAT Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom and NAT
Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE120P in Canada
Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show the status as OK and we have a green light on the card. I'm not sure if the span line is correct. Any help would be greatly appreciated. /etc/zaptel.conf loadzone=us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=24 /etc/asterisk/Zapata.conf [trunkgroups] [channels] language=us context=sasktel switchtype=national signalling=pri_net ; I have also tried pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 channel=1-10 callgroup=1 pickupgroup=1 immediate=no David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] , www.intergraph.com.au http://www.intergraph.com.au ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 650 freezing on Transfer
All, Has anyone experienced a problem with the Polycom 650 phone freezing when you try to do a transfer? I am running asterisk 1.2.20.1 with Polycom SIP version 2.1.1.0052 and boot rom version 3.2.3.0002. I have Polycom 501 phones that work perfectly with the same software versions. The 650 phone; when I hit transfer the caller is placed on hold and the phone is still operational. As soon as I hit a number the phone it will immediately hang and then reboot. Instead of trying to key in a number if I use the KEM for the transfer it works perfectly fine. The problem only occurs when I try to enter a number into the phone and it always freezes on the first digit. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 650 freezing on Transfer
Hi Darren, Thanks for your reply. I have since downgraded to version 2.0.3.0131 and the problem has gone. I am waiting on a link for firmware 2.1.2 so I can try that. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Sent: Friday, 20 July 2007 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 650 freezing on Transfer David, In my experience this is usually due to the 650 being provisioned with a config file that pre-dates SIP version 2.1.1.0052. There's all kinds of things that can lurk in older configs that will cause the newer phone to behave oddly in just the way you describe. New phones need new configs, without exceptions. Try provisioning the 650 with the stock configs supplied with the latest firmware, then go from there. -Darren -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) - Original Message - From: Klaverstyn, David C mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com Sent: Thursday, July 19, 2007 6:34 PM Subject: [asterisk-users] Polycom 650 freezing on Transfer All, Has anyone experienced a problem with the Polycom 650 phone freezing when you try to do a transfer? I am running asterisk 1.2.20.1 with Polycom SIP version 2.1.1.0052 and boot rom version 3.2.3.0002. I have Polycom 501 phones that work perfectly with the same software versions. The 650 phone; when I hit transfer the caller is placed on hold and the phone is still operational. As soon as I hit a number the phone it will immediately hang and then reboot. Instead of trying to key in a number if I use the KEM for the transfer it works perfectly fine. The problem only occurs when I try to enter a number into the phone and it always freezes on the first digit. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CID on Polycom Phones
Hi All, I have a site using Polycom 501 phones and for some reason the caller ID of the phone number is coming up as sip:number@ip of server Does anyone know why? It seems to be a Polycom thing as a Linksys phone displays the CID number as just the number. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID on Polycom Phones
Many thanks, working a beaute. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Tuesday, 17 July 2007 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CID on Polycom Phones Hi David, Disable URL dialing (url-dialing in the feature/ section of sip.cfg. CP Klaverstyn, David C wrote: Hi All, I have a site using Polycom 501 phones and for some reason the caller ID of the phone number is coming up as sip:number@ip of server Does anyone know why? It seems to be a Polycom thing as a Linksys phone displays the CID number as just the number. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hotline with Polycom
Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0:number). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom? The idea is that you pick up the handset and it automatically dials a number. It will be used in a foyer or front door. Many thanks David. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Once Touch Recording
Hi All, Once touch recording only seems to work between extensions. When calling an external party when pressing *1 does nothing. The person you have called can hear 2 DTMF tones. Is there a trick to getting once touch recording working over a zap channel? I am using a TE110P, but calls over SIP to a VSP also fails when trying to use one touch recording. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH Problems.
All, I am using Asterisk 1.4.4 and it is not playing any MOH. I think the underlying problem is the following error: [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player Now it does not matter what I change in the directy= to in the heading [default] in the file musiconhold.conf. [default] mode=files directory=/var/lib/asterisk/moh/klavo I still get the error: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' which does not make sense to me. I don't have any other MOH defined As soon as MOH is initiated is immediately stop with no error. -- Executing [EMAIL PROTECTED]:2] MusicOnHold(SIP/202-0895d428, default) in new stack -- Started music on hold, class 'default', on channel 'SIP/202-0895d428' -- Stopped music on hold on SIP/202-0895d428 It's like the musiconhold.conf file is not read. I have rebooted and reloaded with no chance to the above. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Verizon Interconnection
I have connected with a PRI service with Verizon but not SIP. What is their SIP product as I am not familiar with it? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, 6 June 2007 9:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discussion Subject: [asterisk-users] Re: Verizon Interconnection So absolutely no one here was interconnected with Verizon? I am going to shoot this over to asterisk-biz, also, in hopes someone may have missed it that is on the biz list. The question again is: Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone Installed a Digium TE110P or TE120P card in Canada?
The Telco in Canada is been real painful. I was wondering if anyone has installed a Digium TE1X0P card in Canada and if their Telco was so difficult. The Telco will not provide us a service until they see a FCC or DOC number for the equipment ware are connecting to their service. If have found FCC Part 68, ANSI/ITA-968-A, Including Amendment A1 and A2 Industry Canada CS-03 thanks to Nabeel a forum users. I now need to know if the statement above is what I need to tell the Telco. As I am not in Canada this make it a bit difficult for me. Your help is greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TE120P and Canada FCC or DOC
Hi All, Can anyone tell me if the Digium TE120P card has a FCC or DOC number relevant to Canada? If someone could provide the numbers and or a link to documentation about the numbers it would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LDAPget or something else?
Hi All, We are currently using LDAPget 1.0rc6 with Asterisk 1.2.x. I see that there is LDAPget 2.0rc1 for Asterisk 1.4.x. I was wondering if there was something better. Are people using LDAPget or something else? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 650
All, I have a Polycom 650 phone, when turned on displays Checking application. Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Loudspeaker
Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Loudspeaker
I'm not sure how that could help. At the moment when a call comes in, every phone in the office rings. I would prefer a loudspeaker to ring so it is not in everyone's face so to speak but they are able to hear it in the background. I want to do this for after hour calls and then if no one answers go to a recorded message. The previous system before Asterisk did have a loud speaker that rang so I would prefer to keep it the same if possible. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 16 April 2007 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker Use an ATA to a paging system On 4/15/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Loudspeaker
This is what I want. Do you have any URLs to such a device as I cannot find any. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Monday, 16 April 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with TE110P
Lspci does show: 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) dmesg | tail ip_tables: (C) 2000-2002 Netfilter core team ip_tables: (C) 2000-2002 Netfilter core team tg3: eth0: Link is up at 100 Mbps, full duplex. tg3: eth0: Flow control is on for TX and on for RX. lp: driver loaded but no devices found NET: Registered protocol family 10 Disabled Privacy Extensions on device c0344160(lo) IPv6 over IPv4 tunneling driver divert: not allocating divert_blk for non-ethernet device sit0 eth0: no IPv6 routers present Using CentOS 4.4 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, 2 April 2007 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with TE110P On Mon, Apr 02, 2007 at 02:56:30PM +1000, Klaverstyn, David C wrote: Type in cat /proc/zaptel/* displays Span 1: ZTDUMMY/1 ZTDUMMY/1 1 The driver has not picked up your card. But if I type in lsmod | grep -i wct I get wcte11xp 26016 0 wct4xxp 221120 0 zaptel184996 3 ztdummy,wcte11xp,wct4xxp Two things to do: 1. What is the output of: lspci 2. What really happens when the module loads? rmmod wcte11xp modprobe wcte11xp dmesg | tail What linux distribution is it? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with TE110P
What is the driver for the TE120P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Monday, 2 April 2007 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with TE110P Klaverstyn, David C wrote: Lspci does show: 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) You have a TE120P, not a TE110P, so you are loading the wrong driver module. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with TE110P
From what I can tell the Zaptel 1.2.16 does not have a driver for the TE120P. Is this correct and if so how do I get it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, 3 April 2007 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with TE110P That information is listed in the README file that comes with the Zaptel source code. Klaverstyn, David C wrote: What is the driver for the TE120P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Monday, 2 April 2007 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with TE110P Klaverstyn, David C wrote: Lspci does show: 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) You have a TE120P, not a TE110P, so you are loading the wrong driver module. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with TE110P
OK I now have MODULES:=zaptel wcte12xp In my makefile and the driver loads but the card still fails. lsmod | grep -i wct wcte12xp 39360 0 zaptel184996 1 wcte12xp ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Not sure what this means. Hopefully someone can shed some light. cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE12xP Card 0 IRQ misses: 34 1 WCT1/0/1 2 WCT1/0/2 3 WCT1/0/3 4 WCT1/0/4 5 WCT1/0/5 6 WCT1/0/6 7 WCT1/0/7 8 WCT1/0/8 9 WCT1/0/9 10 WCT1/0/10 11 WCT1/0/11 12 WCT1/0/12 13 WCT1/0/13 14 WCT1/0/14 15 WCT1/0/15 16 WCT1/0/16 17 WCT1/0/17 18 WCT1/0/18 19 WCT1/0/19 20 WCT1/0/20 21 WCT1/0/21 22 WCT1/0/22 23 WCT1/0/23 24 WCT1/0/24 25 WCT1/0/25 26 WCT1/0/26 27 WCT1/0/27 28 WCT1/0/28 29 WCT1/0/29 30 WCT1/0/30 31 WCT1/0/31 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Tuesday, 3 April 2007 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Problems with TE110P From what I can tell the Zaptel 1.2.16 does not have a driver for the TE120P. Is this correct and if so how do I get it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, 3 April 2007 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with TE110P That information is listed in the README file that comes with the Zaptel source code. Klaverstyn, David C wrote: What is the driver for the TE120P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Monday, 2 April 2007 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with TE110P Klaverstyn, David C wrote: Lspci does show: 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) You have a TE120P, not a TE110P, so you are loading the wrong driver module. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with TE110P
OK, Found the problem. It looks like the configuration file is not correct. I added the following line to /etc/sysconfig/zaptel MODULES=$MODULES wcte12xp # TE120P - Single Span T1 Card Once I did this all is now working. Editing the zaptel.sysconfig file in the zaptel source code will also do the same. So I'm guessing anyone with a TE120P card will need to do the same until Asterisk update the files for the TE120P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo Cordeiro Sent: Tuesday, 3 April 2007 10:11 AM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Problems with TE110P Permissions? Sds, Gustavo From: Klaverstyn, David C [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [asterisk-users] Problems with TE110P Date: Tue, 3 Apr 2007 09:59:19 +1000 OK I now have MODULES:=zaptel wcte12xp In my makefile and the driver loads but the card still fails. lsmod | grep -i wct wcte12xp 39360 0 zaptel184996 1 wcte12xp ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Not sure what this means. Hopefully someone can shed some light. cat /proc/zaptel/* Span 1: WCT1/0 Wildcard TE12xP Card 0 IRQ misses: 34 1 WCT1/0/1 2 WCT1/0/2 3 WCT1/0/3 4 WCT1/0/4 5 WCT1/0/5 6 WCT1/0/6 7 WCT1/0/7 8 WCT1/0/8 9 WCT1/0/9 10 WCT1/0/10 11 WCT1/0/11 12 WCT1/0/12 13 WCT1/0/13 14 WCT1/0/14 15 WCT1/0/15 16 WCT1/0/16 17 WCT1/0/17 18 WCT1/0/18 19 WCT1/0/19 20 WCT1/0/20 21 WCT1/0/21 22 WCT1/0/22 23 WCT1/0/23 24 WCT1/0/24 25 WCT1/0/25 26 WCT1/0/26 27 WCT1/0/27 28 WCT1/0/28 29 WCT1/0/29 30 WCT1/0/30 31 WCT1/0/31 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Tuesday, 3 April 2007 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Problems with TE110P From what I can tell the Zaptel 1.2.16 does not have a driver for the TE120P. Is this correct and if so how do I get it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, 3 April 2007 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with TE110P That information is listed in the README file that comes with the Zaptel source code. Klaverstyn, David C wrote: What is the driver for the TE120P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Monday, 2 April 2007 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with TE110P Klaverstyn, David C wrote: Lspci does show: 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) You have a TE120P, not a TE110P, so you are loading the wrong driver module. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Verificador de Segurança do Windows Live OneCare: combata já vírus e outras ameaças! http://onecare.live.com/site/pt-br/default.htm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE120P and Unknown Signalling Method
I have a brand new TE120P card that I have installed and asterisk is not starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown signalling method 'pri_cpe' It seems it does not matter what I change the vaule for signalling= to, it always returns it as invalid. I have tried the config from my other 2 servers running TE110P cards and the config from AusTechPartniships and they always come back as the error above. The drivers seem to be loaded ok: lsmod wcte12xp393600 zaptel184996 1 wcte12xp crc_ccitt 6209 1 zaptel ztcfg -v Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 11 channels configured. Does anyone have any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with TE110P
I have a new server using Zaptel 1.2.16 Issuing a ztcfg gives the following error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) Issuing the command lsmod | grep -i wct results in: wcte11xp 26016 0 wct4xxp 221120 0 zaptel184996 3 ztdummy,wcte11xp,wct4xxp In my zaptel Makefile I have: MODULES:=zaptel wcte11xp Please help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with TE110P
I forgot to mention that my /etc/zaptel.conf file contains: loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 I have a new server using Zaptel 1.2.16 Issuing a ztcfg gives the following error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) Issuing the command lsmod | grep -i wct results in: wcte11xp 26016 0 wct4xxp 221120 0 zaptel184996 3 ztdummy,wcte11xp,wct4xxp In my zaptel Makefile I have: MODULES:=zaptel wcte11xp Please help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with TE110P
Type in cat /proc/zaptel/* displays Span 1: ZTDUMMY/1 ZTDUMMY/1 1 But if I type in lsmod | grep -i wct I get wcte11xp 26016 0 wct4xxp 221120 0 zaptel184996 3 ztdummy,wcte11xp,wct4xxp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, 2 April 2007 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problems with TE110P On Mon, Apr 02, 2007 at 12:50:18PM +1000, Klaverstyn, David C wrote: I have a new server using Zaptel 1.2.16 Issuing a ztcfg gives the following error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 Now what do you actually have loaded? cat /proc/zaptel/* -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy and MOH
Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI zap show status Description Alarms IRQbpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct. Please help. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ztdummy and MOH
WOW that fixed it! What an Idiot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 28 March 2007 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ztdummy and MOH On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. If you rub your hand across the mouthpiece of the phone, does the music play? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ztdummy and MOH
The cli shows: -- Started music on hold, class 'jessica', on channel 'IAX2/205-3' -- Stopped music on hold on IAX2/205-3 I am using MP3 but I also tried it with WAV and GSM with the same result. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 28 March 2007 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ztdummy and MOH On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: WOW that fixed it! What an Idiot. I was going somewhere with that, but never mind. Good luck. Maybe the idiot is the guy who posted no additional details of his configuration, in particular, whether the CLI was showing music on hold starting, and then stopping, or whether the music on hold process was continuing but no sound. If it was a timing issue, by rubbing your hand across the mouthpiece, I would guess it is generating interupts for the timer to work and music on hold works, until you stop rubbing it and it fades it out. Hitting or tapping the mouthpiece produces the same outcome. Or, it that doesn't produce anything, it could be a permissions problem. It could be something not configured correctly in the config file. It could be that you are using mp3s instead of native format, as Andrew had asked about. But, since I'm an idiot, what do I know? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ztdummy and MOH
I am using autoload and I have rebooted the server. I have tried using different files and a different location. This is getting very frustrating. I wish I knew what the problem was. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 28 March 2007 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ztdummy and MOH On 3/28/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: The cli shows: -- Started music on hold, class 'jessica', on channel 'IAX2/205-3' -- Stopped music on hold on IAX2/205-3 That rules out the timing. I see this note in the config file: ; If you are not using autoload in modules.conf, then you ; must ensure that the format modules for any formats you wish ; to use are loaded _before_ res_musiconhold. If you do not do ; this, res_musiconhold will skip the files it is not able to ; understand when it loads. Does that apply? Also, I'm not sure if this still applies, but at one time, you had to issue a restart command if you added any music files for the Asterisk to see them. A reload command wouldn't do it. Have you tried restart (not of the system, just Asterisk from cli). Another thing you may or not be able to check... what if you just put the files in the default directory and in the default context? Do they work then? This would eliminate some of the musiconhold config options causing problems. I guess along those lines do the default music on hold files work? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Back
Can you provide some specific details as I would like to implement something like this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov Sent: Tuesday, 13 March 2007 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Back In general how I implemented is as follows: - Caller calls asterisk. - From AGI, asterisk gets the caller ID. - Without answering, play back beeps, to simulate busy. - When the user hangs up, asterisk detects broken connection, cannot send beeps, and Originates a call back to the caller. - After the caller hangs up, the Originate from Asterisk rings his phone. - As soon as he picks up, asterisk ask him to enter destination. - The caller enter destination #, asterisk dials the destination. - The two channels are connected. That's it. Works great. Luis Claudio Santos wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Back
This doesn't make sense to me. Are you able to give some example dial plans? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov Sent: Tuesday, 13 March 2007 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Back All the code is in AGI. Take a look at the Originate application. (http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Actio n+Originate) Klaverstyn, David C wrote: Can you provide some specific details as I would like to implement something like this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov Sent: Tuesday, 13 March 2007 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Back In general how I implemented is as follows: - Caller calls asterisk. - From AGI, asterisk gets the caller ID. - Without answering, play back beeps, to simulate busy. - When the user hangs up, asterisk detects broken connection, cannot send beeps, and Originates a call back to the caller. - After the caller hangs up, the Originate from Asterisk rings his phone. - As soon as he picks up, asterisk ask him to enter destination. - The caller enter destination #, asterisk dials the destination. - The two channels are connected. That's it. Works great. Luis Claudio Santos wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] gtalk2voip and Asterisk
After upgrading to Asterisk 1.4.1 from 1.4.0 it just worked for me. There must have been a bug in 1.4.0. I have successfully connected to a Gmail and MSN instant message client. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mani Sridhar Sent: Saturday, 3 March 2007 8:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] gtalk2voip and Asterisk hi, i was able to get this working with google talk. i entered [EMAIL PROTECTED] using the gtalk2voip.com website's invite box, and as a result, saw a request from [EMAIL PROTECTED] to be added as a buddy in my google talk contact list. i accepted the request. in my asterisk dialplan, i have this entry... exten = 3501, 1, Dial(SIP/[EMAIL PROTECTED]) this allows any extension in my asterisk box to dial 3501, and [EMAIL PROTECTED] receives the call on the google talk client. the call is established, and voice quality is good. to allow a call from google talk client for [EMAIL PROTECTED], i opened a chat window to the buddy [EMAIL PROTECTED] and typed call [EMAIL PROTECTED], and this made extension ABC on my asterisk box start ringing. again, the call was established, and audio was ok. as far as asterisk is concerned, this is a SIP call. bottomline - it's a good alternative to using the native jabber/jingle library in asterisk 1.4 . in fact, i haven't been able to get asterisk to successfully set up a call to googletalk using the chan_gtalk module . i am inside a NAT-ed LAN, and audio works in one direction only for the asterisk (SIP) - gtalk call. anyone else got asterisk - googletalk using chan_gtalk working? Message: 10 Date: Fri, 02 Mar 2007 19:07:41 +0200 From: Cosmin Prund [EMAIL PROTECTED] Subject: Re: [asterisk-users] gtalktovoip and Asteirsk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I don't think it works. I tried calling my own yahoo messenger ID with no success: it rings a number of times and then it goes to some sort of voice mail. And I did invite the user they specified to my yahoo list, I also entered my yahoo id into the registration form on the site. I used a extensions.conf command like this for the try: exten = 641,1,Dial(SIP/[EMAIL PROTECTED]) (and yes, that's one of the yahoo ID I tryed with, and I don't think it exists! ) Klaverstyn, David C wrote: Has anyone managed to get gtalktovoip working at all? If so please explain. http://www.gtalk2voip.com/faq.shtml *2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ?* A: This is a major feature of our gateway and it is very easy. oGTalk: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] oMSN: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] oYahoo: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- _ Get Married in 2007. Join Shaadi.com http://www.shaadi.com/ptnr.php?ptnr=mhottag ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gtalktovoip and Asteirsk
Has anyone managed to get gtalktovoip working at all? If so please explain. http://www.gtalk2voip.com/faq.shtml 2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ? A: This is a major feature of our gateway and it is very easy. oGTalk: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] oMSN: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] oYahoo: [EMAIL PROTECTED] can be reached by calling to sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Not registering Port with VSP
All, I'm guessing no one knows the answer as to why when I register with a VSP I am not sending a Port number with the registration but only my IP address. If anyone has any answers it would be greatly appreciated. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Wednesday, 28 February 2007 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not registering Port with VSP Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been told that my asterisk server is registering my IP with the VSP but the port is empty. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not registering Port with VSP
Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been told that my asterisk server is registering my IP with the VSP but the port is empty. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto load of zap drivers
My understanding is that with Asterisk 1.2.x issuing the command of make install-udev allowed the drivers to be loaded upon the server boot. Doing this with version 1.4 does not seem to work. Using menuselect I selected zaptel and ztdummy. Should I also be selecting something else for the drivers to load at start up? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Auto load of zap drivers
I am running CentOS 4.4. You say I need modprobe ztdummy on startup. I though the udev option made that happen. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, 20 February 2007 3:11 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Auto load of zap drivers On Tue, Feb 20, 2007 at 11:05:31AM +1100, Klaverstyn, David C wrote: My understanding is that with Asterisk 1.2.x issuing the command of make install-udev allowed the drivers to be loaded upon the server boot. Doing this with version 1.4 does not seem to work. Those udev rules are responsible for the generation of files under /dev/zap/ . Which distribution do you use? Using menuselect I selected zaptel and ztdummy. Should I also be selecting something else for the drivers to load at start up? All you need to run on startup is: modprobe ztdummy Nothing more. Not even a ztcfg. The zaptel init script tries doing that if it senses you have no other zaptel timing source. Do you have ztdummy and zaptel available? modinfo ztdummy modinfo zaptel -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Useragent List
All, Is there a way I can get a list of all my users' useragents? I basically want to know what firmware each of my phones have without having to list them individually. Thanks David. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400 with 1 FXO
Hi All, I cannot get my TDM to work correctly. In my /etc/zaptel.conf file I have loadzone = us defaultzone=us fxoks=1 In my /etc/asterisk/zapata.conf file I have [trunkgroups] [channels] context=from-pstn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel = 1 running a ztcfg -vv give the following Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. A reload of asterisk give the following error: Feb 8 18:38:43 ERROR[4558]: chan_zap.c:10305 setup_zap: Unable to reconfigure channel '1' Feb 8 18:38:43 WARNING[4558]: chan_zap.c:11067 reload: Reload of chan_zap.so is unsuccessful! Any help would be greatly appreciated. For what it is worth this system is located in Canada. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400 with 1 FXO
Hi, Yes it should, I have changed it back and is still causing the same problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Friday, 9 February 2007 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400 with 1 FXO Klaverstyn, David C wrote: Hi All, I cannot get my TDM to work correctly. In my /etc/zaptel.conf file I have loadzone = us defaultzone=us fxoks=1 Shouldn't this be fxsks if you're using an FXO module as analog trunk? Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400 with 1 FXO
Yes, I have also since put that in and I get the error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring signalling And if I put in rxwink I get this error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring rxwink It's all very strange. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Friday, 9 February 2007 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400 with 1 FXO Klaverstyn, David C wrote: Hi, Yes it should, I have changed it back and is still causing the same problems. Did you also missed out the following line in zapata.conf? signalling=fxs_ks Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400 with 1 FXO
My original post does have the contents of the file exactly. In my /etc/asterisk/zapata.conf file I have [trunkgroups] [channels] context=from-pstn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Friday, 9 February 2007 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400 with 1 FXO Klaverstyn, David C wrote: Yes, I have also since put that in and I get the error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring signalling And if I put in rxwink I get this error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring rxwink It's all very strange. please post your complete zapata.conf - I think there's a preceding line that's confusing the parser. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400 with 1 FXO
Hi, I stuffed Up, Here is my correct contents ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes busydetect=yes busycount=6 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no channel = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, 9 February 2007 2:29 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] TDM400 with 1 FXO From: Klaverstyn, David C [EMAIL PROTECTED] Date: Fri, 9 Feb 2007 15:12:49 +1100 My original post does have the contents of the file exactly. You haven't defined channel signaling in zapata.conf. Need something like signalling = fxs_ks according to your zaptel.conf. Yuan Liu In my /etc/asterisk/zapata.conf file I have [trunkgroups] [channels] context=from-pstn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Friday, 9 February 2007 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400 with 1 FXO Klaverstyn, David C wrote: Yes, I have also since put that in and I get the error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring signalling And if I put in rxwink I get this error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring rxwink It's all very strange. please post your complete zapata.conf - I think there's a preceding line that's confusing the parser. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
