Re: [asterisk-users] Zaptel problem with pots lines
Hi Enrico, have you tried with busydetect=yes? It (sometimes) worked for me with Asterisk 1.2. Giorgio Enrico Maistro wrote: Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With callprogress=no line get answered as soon as it start ringing, regardless if someone really answer the call. Zaptel channels use fxs_ks signalling . Loading wctdm module with debug=1 result in: kernel: Freshmaker version: 73 kernel: Freshmaker passed register test kernel: ProSLIC on module 0, product 3, version 15 kernel: VoiceDAA System: 04 kernel: ISO-Cap is now up, line side: 03 rev 06 kernel: setting FXO tx gain for card=0 to 0 kernel: setting FXO rx gain for card=0 to 0 kernel: DEBUG fxotxgain:0.0 fxorxgain:0.0 kernel: Module 0: Installed -- AUTO FXO (FCC mode) kernel: ProSLIC on module 1, product 0, version 0 kernel: VoiceDAA System: 04 kernel: ISO-Cap is now up, line side: 03 rev 06 kernel: setting FXO tx gain for card=1 to 0 kernel: setting FXO rx gain for card=1 to 0 kernel: DEBUG fxotxgain:0.0 fxorxgain:0.0 kernel: Module 1: Installed -- AUTO FXO (FCC mode) kernel: ProSLIC on module 2, product 0, version 0 kernel: Module 2: Not installed kernel: ProSLIC on module 3, product 0, version 0 kernel: Module 3: Not installed kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) kernel: 4294908325 Polarity reversed (0 - -1) kernel: 4294908326 Polarity reversed (0 - 1) kernel: BATTERY on 1/1 (-)! kernel: NO BATTERY on 1/2! Any suggestion? Am i trying to do something that simply can't be done? Thanks, Enrico Maistro ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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] Asterisk unable to register to tnet.it
Hi Marino, I tried to connect zoiper directly to the provider with the same account parameters I'm using with Asterisk. Zoiper connects without problems. It is true tnet.it is not resolvable but I can use the proxy URL sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm trying to understand where is the problem. I thought I had to specify the outboundproxy parameter in the general section of sip.conf to make Asterisk correctly work but it seems that's not enough. Thank you. Giorgio map wrote: Hi Giorgio, From your email seems clear that your Asterisk box can not resolve tnet.it http://tnet.it and SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, Asterisk gives a timeout on registration and a no such host because cannot resolve tnet.it http://tnet.it but that server address is not resolvable so I think that is not a problem (my zoiper connects to the provider without problems, so why shouldn't Asterisk??) Activating sip debug shows the register packets but nothing in return. I used the proxy tnet gave me but nothing changes. Searched on their site for some help about Asterisk configuration but nothing...the same on the rest of internet. Giorgio map wrote: Hi Giorgio, Do you have any log showing some error? Did you already have a look at SIP connection messages from and to this SIP server? I suggest you to use wireshark to check sip messages. Thanks, Marino On Mon, Jul 14, 2008 at 3:47 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I cannot make my Asterisk register to tnet.it http://tnet.it http://tnet.it, an italian SIP provider. I tried many register string formats and tried to set realm and outboundproxy (sip.tnet.it http://sip.tnet.it http://sip.tnet.it) too but without any result. Still I cannot register (but for example messagenet works fine). Is there anybody who tried this provider and successfully registered to it? Thank you. Giorgio. ___ -- 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 ___ -- 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] Asterisk unable to register to tnet.it
Hi Giorgio, Just to recap: 1) you are able to connect to tnet.it by using the same account of your asterisk box. There is no issue related to your account. 2) Could you please confirm that you are running zoiper from the same box used by asterisk? If yes we can exclude some generic network issues. From your previous email : ... Activating sip debug shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named Ethereal in order to check sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Marino, I tried to connect zoiper directly to the provider with the same account parameters I'm using with Asterisk. Zoiper connects without problems. It is true tnet.it is not resolvable but I can use the proxy URL sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm trying to understand where is the problem. I thought I had to specify the outboundproxy parameter in the general section of sip.conf to make Asterisk correctly work but it seems that's not enough. Thank you. Giorgio map wrote: Hi Giorgio, From your email seems clear that your Asterisk box can not resolve tnet.it http://tnet.it and SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, Asterisk gives a timeout on registration and a no such host because cannot resolve tnet.it http://tnet.it but that server address is not resolvable so I think that is not a problem (my zoiper connects to the provider without problems, so why shouldn't Asterisk??) Activating sip debug shows the register packets but nothing in return. I used the proxy tnet gave me but nothing changes. Searched on their site for some help about Asterisk configuration but nothing...the same on the rest of internet. Giorgio map wrote: Hi Giorgio, Do you have any log showing some error? Did you already have a look at SIP connection messages from and to this SIP server? I suggest you to use wireshark to check sip messages. Thanks, Marino On Mon, Jul 14, 2008 at 3:47 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I cannot make my Asterisk register to tnet.it http://tnet.it http://tnet.it, an italian SIP provider. I tried many register string formats and tried to set realm and outboundproxy (sip.tnet.it http://sip.tnet.it http://sip.tnet.it) too but without any result. Still I cannot register (but for example messagenet works fine). Is there anybody who tried this provider and successfully registered to it? Thank you. Giorgio. ___ -- 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
Re: [asterisk-users] Zaptel problem with pots lines
Hi Enrico. In Italy the polarity reversal is never used. I'm using the TDM400 with an FXO port in Italy with the config reported below and is working properly in any situations: --- zaptel.conf --- fxsks=1 loadzone=it defaultzone=it --- zapata.conf --- [channels] language=en context=from-tdm-fxo signalling=fxs_ks threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes busydetect=yes busycount=6 usecallerid=yes callerid=asreceived echocancel=yes echocancelwhenbridged=no relaxdtmf=yes rxgain=3.0 txgain=0.0 amaflags=billing group=1 callgroup=1 pickupgroup=1 jbenable=yes faxdetect=no channel = 1 I'm using zaptel-1.4.6 and asterisk-1.4.20.1. I hope this could help you. Best regards, Marco Signorini. Enrico Maistro wrote: Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With callprogress=no line get answered as soon as it start ringing, regardless if someone really answer the call. Zaptel channels use fxs_ks signalling . Loading wctdm module with debug=1 result in: kernel: Freshmaker version: 73 kernel: Freshmaker passed register test kernel: ProSLIC on module 0, product 3, version 15 kernel: VoiceDAA System: 04 kernel: ISO-Cap is now up, line side: 03 rev 06 kernel: setting FXO tx gain for card=0 to 0 kernel: setting FXO rx gain for card=0 to 0 kernel: DEBUG fxotxgain:0.0 fxorxgain:0.0 kernel: Module 0: Installed -- AUTO FXO (FCC mode) kernel: ProSLIC on module 1, product 0, version 0 kernel: VoiceDAA System: 04 kernel: ISO-Cap is now up, line side: 03 rev 06 kernel: setting FXO tx gain for card=1 to 0 kernel: setting FXO rx gain for card=1 to 0 kernel: DEBUG fxotxgain:0.0 fxorxgain:0.0 kernel: Module 1: Installed -- AUTO FXO (FCC mode) kernel: ProSLIC on module 2, product 0, version 0 kernel: Module 2: Not installed kernel: ProSLIC on module 3, product 0, version 0 kernel: Module 3: Not installed kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) kernel: 4294908325 Polarity reversed (0 - -1) kernel: 4294908326 Polarity reversed (0 - 1) kernel: BATTERY on 1/1 (-)! kernel: NO BATTERY on 1/2! Any suggestion? Am i trying to do something that simply can't be done? Thanks, Enrico Maistro ___ ___ -- 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] Asterisk unable to register to tnet.it
Hi Marino, 1) yes I can connect using the account 2) no, I'm running zoiper on a different machine. I'm using an Asterisk server which is not behind nat as for the machine zoiper is runnin' on. The Asterisk server is directly connected to internet, I wanted to avoid nat problems, that's why. Moreover I tried to create a simpler account on my zoiper using username, password and domain name only and it works even without setting the sip proxy. I changed the Asterisk server too: now I'm using a test one where I can ping tnet.it from... but nothing changes. I'm using this string: register = 0442410280:provapolika:[EMAIL PROTECTED]/0442410280 I changed it in many other forms following the wiki pages but nothing. I see sip packets are sent to tnet.it (I set up sip debug) but I always get this message: Jul 15 10:06:39 NOTICE[3281]: chan_sip.c:5495 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) I wonder why I had no problems with the other provider we are using while tnet.it is making me get crazy Thank you. Giorgio map wrote: Hi Giorgio, Just to recap: 1) you are able to connect to tnet.it http://tnet.it by using the same account of your asterisk box. There is no issue related to your account. 2) Could you please confirm that you are running zoiper from the same box used by asterisk? If yes we can exclude some generic network issues. From your previous email : ... Activating sip debug shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named Ethereal in order to check sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, I tried to connect zoiper directly to the provider with the same account parameters I'm using with Asterisk. Zoiper connects without problems. It is true tnet.it http://tnet.it is not resolvable but I can use the proxy URL sip.tnet.it http://sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm trying to understand where is the problem. I thought I had to specify the outboundproxy parameter in the general section of sip.conf to make Asterisk correctly work but it seems that's not enough. Thank you. Giorgio map wrote: Hi Giorgio, From your email seems clear that your Asterisk box can not resolve tnet.it http://tnet.it http://tnet.it and SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, Asterisk gives a timeout on registration and a no such host because cannot resolve tnet.it http://tnet.it http://tnet.it but that server address is not resolvable so I think that is not a problem (my zoiper connects to the provider without problems, so why shouldn't Asterisk??) Activating sip debug shows the register packets but nothing in return. I used the proxy tnet gave me but nothing changes. Searched on their site for some help about Asterisk configuration but nothing...the same on the rest of internet. Giorgio map wrote: Hi Giorgio, Do you have any log showing some error? Did you already have a look at SIP connection messages from and to this SIP server? I suggest you to use wireshark to check sip messages. Thanks, Marino On Mon, Jul 14, 2008 at 3:47 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi,
Re: [asterisk-users] How to integerate 2 TDM cards on same machine.
Thanks Noah. It is now properly running. Thanks again regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Tuesday, July 15, 2008 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to integerate 2 TDM cards on same machine. Hi Syed - zttool shows that TDM800P is loaded first and TDM2401E is loaded second. now problem is ports are not being configured by asterisk. i have done following changes in two files zaptel.onf and zapata.conf. zaptel.conf loadzone=us, defaultzone=us, fxoks=1-4, fxsks=5-8, fxsks=9-32(or should this be fxoks???) zapata.conf signalling=fxoks channels =1-4 signalling=fxsks channels = 5-8 signalling=fxsks channels = 9-32 please see the bold lines. since FXO ports use FXS signalling so i used fxsks. is this right or wrong. are these changes have to be made in both the files as i have done or only in zaptel.conf waiting for information Almost there. Your zaptel.conf is correct (sorry I gave you the wrong signalling before). In zapata.conf, your signalling lines should look like: signalling=fxo_ks channels = 1-4 signalling=fxs_ks channels = 5-32 - Noah ___ -- 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] Asterisk unable to register to tnet.it
Hi Giorgio, RE my point 2: You should test a sip client, whatever you want, on your linux/asterisk box just to double check that this box works fine. If you are abel to connect with a sip client from tour asterisk box we will be sure that the network configuration is ok. You have no natt but maybe your routing table is not correct :-) Do you already test to just ping to tnet.it port 5060 ? Marino On Tue, Jul 15, 2008 at 10:27 AM, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Marino, 1) yes I can connect using the account 2) no, I'm running zoiper on a different machine. I'm using an Asterisk server which is not behind nat as for the machine zoiper is runnin' on. The Asterisk server is directly connected to internet, I wanted to avoid nat problems, that's why. Moreover I tried to create a simpler account on my zoiper using username, password and domain name only and it works even without setting the sip proxy. I changed the Asterisk server too: now I'm using a test one where I can ping tnet.it from... but nothing changes. I'm using this string: register = 0442410280:provapolika:[EMAIL PROTECTED]/0442410280 I changed it in many other forms following the wiki pages but nothing. I see sip packets are sent to tnet.it (I set up sip debug) but I always get this message: Jul 15 10:06:39 NOTICE[3281]: chan_sip.c:5495 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) I wonder why I had no problems with the other provider we are using while tnet.it is making me get crazy Thank you. Giorgio map wrote: Hi Giorgio, Just to recap: 1) you are able to connect to tnet.it http://tnet.it by using the same account of your asterisk box. There is no issue related to your account. 2) Could you please confirm that you are running zoiper from the same box used by asterisk? If yes we can exclude some generic network issues. From your previous email : ... Activating sip debug shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named Ethereal in order to check sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, I tried to connect zoiper directly to the provider with the same account parameters I'm using with Asterisk. Zoiper connects without problems. It is true tnet.it http://tnet.it is not resolvable but I can use the proxy URL sip.tnet.it http://sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm trying to understand where is the problem. I thought I had to specify the outboundproxy parameter in the general section of sip.conf to make Asterisk correctly work but it seems that's not enough. Thank you. Giorgio map wrote: Hi Giorgio, From your email seems clear that your Asterisk box can not resolve tnet.it http://tnet.it http://tnet.it and SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, Asterisk gives a timeout on registration and a no such host because cannot resolve tnet.it http://tnet.it http://tnet.it but that server address is not resolvable so I think that is not a problem (my zoiper connects to the provider without problems, so why shouldn't Asterisk??) Activating sip debug shows the register packets but nothing in return. I used the proxy tnet gave me but nothing changes. Searched on their site for some help about Asterisk configuration but nothing...the same on the rest of internet. Giorgio map wrote: Hi Giorgio, Do you have any log showing some error? Did you
Re: [asterisk-users] Asterisk unable to register to tnet.it
Check dns server entries in asterisk box . /etc/resolv.conf . Put opendns servers ip there just to test . opendns ip's are 208.67.220.220 and 208.67.222.222 On Tue, Jul 15, 2008 at 2:19 PM, map [EMAIL PROTECTED] wrote: Hi Giorgio, RE my point 2: You should test a sip client, whatever you want, on your linux/asterisk box just to double check that this box works fine. If you are abel to connect with a sip client from tour asterisk box we will be sure that the network configuration is ok. You have no natt but maybe your routing table is not correct :-) Do you already test to just ping to tnet.it port 5060 ? Marino On Tue, Jul 15, 2008 at 10:27 AM, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Marino, 1) yes I can connect using the account 2) no, I'm running zoiper on a different machine. I'm using an Asterisk server which is not behind nat as for the machine zoiper is runnin' on. The Asterisk server is directly connected to internet, I wanted to avoid nat problems, that's why. Moreover I tried to create a simpler account on my zoiper using username, password and domain name only and it works even without setting the sip proxy. I changed the Asterisk server too: now I'm using a test one where I can ping tnet.it from... but nothing changes. I'm using this string: register = 0442410280:provapolika:[EMAIL PROTECTED]/0442410280 I changed it in many other forms following the wiki pages but nothing. I see sip packets are sent to tnet.it (I set up sip debug) but I always get this message: Jul 15 10:06:39 NOTICE[3281]: chan_sip.c:5495 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) I wonder why I had no problems with the other provider we are using while tnet.it is making me get crazy Thank you. Giorgio map wrote: Hi Giorgio, Just to recap: 1) you are able to connect to tnet.it http://tnet.it by using the same account of your asterisk box. There is no issue related to your account. 2) Could you please confirm that you are running zoiper from the same box used by asterisk? If yes we can exclude some generic network issues. From your previous email : ... Activating sip debug shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named Ethereal in order to check sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, I tried to connect zoiper directly to the provider with the same account parameters I'm using with Asterisk. Zoiper connects without problems. It is true tnet.it http://tnet.it is not resolvable but I can use the proxy URL sip.tnet.it http://sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm trying to understand where is the problem. I thought I had to specify the outboundproxy parameter in the general section of sip.conf to make Asterisk correctly work but it seems that's not enough. Thank you. Giorgio map wrote: Hi Giorgio, From your email seems clear that your Asterisk box can not resolve tnet.it http://tnet.it http://tnet.it and SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, Asterisk gives a timeout on registration and a no such host because cannot resolve tnet.it http://tnet.it http://tnet.it but that server address is not resolvable so I think that is not a problem (my zoiper connects to the provider without problems, so why shouldn't Asterisk??) Activating sip debug shows the register packets but nothing in return. I used the proxy tnet gave me but nothing changes. Searched on their site for some help about Asterisk
Re: [asterisk-users] Zaptel problem with pots lines
Hi Noah, Hi Enrico - I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. Zaptel channels use fxs_ks signalling . I must admit I know nothing about Italian phone lines, but maybe you could try other signalling methods? Maybe ground start or loop start would work. - Noah Unfortunatly i already tried both loop start and ground start without any luck... after googling around for a while it seems clear that the right signalling to use here in Italy is ks. Enrico ___ -- 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] Zaptel problem with pots lines
Hi Giorgio, Giorgio Incantalupo wrote: Hi Enrico, have you tried with busydetect=yes? It (sometimes) worked for me with Asterisk 1.2. Giorgio I'm already using busydetect=yes to detect hangup and busy conditions with good results, but it doesn't seem to be of any help on detecting answer conditions. Enrico ___ -- 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] Zaptel problem with pots lines
Hi Marco, Marco Signorini wrote: Hi Enrico. In Italy the polarity reversal is never used. Good to know... at least i can stop messing with it. I'm using the TDM400 with an FXO port in Italy with the config reported below and is working properly in any situations: --- zaptel.conf --- fxsks=1 loadzone=it defaultzone=it My zaptel.conf is exactly the same. --- zapata.conf --- [channels] language=en context=from-tdm-fxo signalling=fxs_ks threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes busydetect=yes busycount=6 usecallerid=yes callerid=asreceived echocancel=yes echocancelwhenbridged=no relaxdtmf=yes rxgain=3.0 txgain=0.0 amaflags=billing group=1 callgroup=1 pickupgroup=1 jbenable=yes faxdetect=no channel = 1 My zapata.conf differs in: language = it instead of en rxgain = 0.0 instead of 3.0 jbenable = no instead of yes Unfortunatly even with your exact same configuration nothing change. I'm using zaptel-1.4.6 and asterisk-1.4.20.1. I'll give a try with zaptel-1.4.6... I hope this could help you. Best regards, Marco Signorini. Regards, Enrico Maistro ___ -- 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] Zaptel problem with pots lines
Hi Enrico. I'm quite sure that the differences you have in the zapata.conf doesn't have any effect on the problem. If I'm not wrong: language=it tells asterisk to use the italian sounds (if available) for any calls related to this zap channel; rxgain = 0.0 is related only to perceived audio gain jbenable = no is forcing the jitter buffer off for this zap channel. Let us know if you have problems with zaptel-1.4.6. I can assure that with this version and this configuration more than one installation with TDM400P I'm responsible for is working fine (since 1.4.6 came out). Could be that the problem is related to Asterisk 1.6? Unfortunately I never had the possibility to try this new version. Best regards, Marco Signorini. Enrico Maistro wrote: My zapata.conf differs in: language = it instead of en rxgain = 0.0 instead of 3.0 jbenable = no instead of yes Unfortunatly even with your exact same configuration nothing change. I'm using zaptel-1.4.6 and asterisk-1.4.20.1. I'll give a try with zaptel-1.4.6... ___ -- 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] Zaptel problem with pots lines
On Mon, Jul 14, 2008 at 08:56:40PM +0200, Enrico Maistro wrote: Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. AFAIK chan_zap can only detect answer if it is provided through a polarity reversal. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Asterisk unable to register to tnet.it
Hi all, I solved it I tried with an Asterisk 1.4 test box. It said: ast_get_srv: SRV lookup for '_sip._udp.tnet.it' mapped to host sip.tnet.it, port 5060 and...it seems to work!! So I put srvlookup=yes on Asterisk 1.2 and IT WORKS!!! Now I try to make calls. Thank you all for patience!! Giorgio Jaswinder Singh wrote: Check dns server entries in asterisk box . /etc/resolv.conf . Put opendns servers ip there just to test . opendns ip's are 208.67.220.220 and 208.67.222.222 On Tue, Jul 15, 2008 at 2:19 PM, map [EMAIL PROTECTED] wrote: Hi Giorgio, RE my point 2: You should test a sip client, whatever you want, on your linux/asterisk box just to double check that this box works fine. If you are abel to connect with a sip client from tour asterisk box we will be sure that the network configuration is ok. You have no natt but maybe your routing table is not correct :-) Do you already test to just ping to tnet.it port 5060 ? Marino On Tue, Jul 15, 2008 at 10:27 AM, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Marino, 1) yes I can connect using the account 2) no, I'm running zoiper on a different machine. I'm using an Asterisk server which is not behind nat as for the machine zoiper is runnin' on. The Asterisk server is directly connected to internet, I wanted to avoid nat problems, that's why. Moreover I tried to create a simpler account on my zoiper using username, password and domain name only and it works even without setting the sip proxy. I changed the Asterisk server too: now I'm using a test one where I can ping tnet.it from... but nothing changes. I'm using this string: register = 0442410280:provapolika:[EMAIL PROTECTED]/0442410280 I changed it in many other forms following the wiki pages but nothing. I see sip packets are sent to tnet.it (I set up sip debug) but I always get this message: Jul 15 10:06:39 NOTICE[3281]: chan_sip.c:5495 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) I wonder why I had no problems with the other provider we are using while tnet.it is making me get crazy Thank you. Giorgio map wrote: Hi Giorgio, Just to recap: 1) you are able to connect to tnet.it http://tnet.it by using the same account of your asterisk box. There is no issue related to your account. 2) Could you please confirm that you are running zoiper from the same box used by asterisk? If yes we can exclude some generic network issues. From your previous email : ... Activating sip debug shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named Ethereal in order to check sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, I tried to connect zoiper directly to the provider with the same account parameters I'm using with Asterisk. Zoiper connects without problems. It is true tnet.it http://tnet.it is not resolvable but I can use the proxy URL sip.tnet.it http://sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm trying to understand where is the problem. I thought I had to specify the outboundproxy parameter in the general section of sip.conf to make Asterisk correctly work but it seems that's not enough. Thank you. Giorgio map wrote: Hi Giorgio, From your email seems clear that your Asterisk box can not resolve tnet.it http://tnet.it http://tnet.it and SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, Asterisk gives a timeout on registration and a no such host because cannot resolve tnet.it http://tnet.it http://tnet.it but that server address is not resolvable so I think that is not a problem (my zoiper connects to the provider without
Re: [asterisk-users] zap not getting callerid any more
On Sun, 2008-07-13 at 10:22 -0400, Brian J. Murrell wrote: I have a wildcard 100 xp on my pots line and all was working just fine up until a few days ago when all of a sudden it stopped receiving caller id on incoming calls. I know caller id is being presented on the line as the analog set on the same line always gets it. ... rxgain=10.9 txgain=0.0 To add more information, I tried setting my rxgain back to 0.0 and CID now seems more reliable. So again I went through the process of setting rxgain using a milliwatt number (not at all local however as I can't find one near me) and ztmonitor and had to adjust it down by 4 points to 10.5 to get it close to the magical 14844. CID was again, 100% unreliable. One thing I have noticed is that in the cases where the wildcard cannot determine the CID (i.e. because the rxgain is up around 10.5), I get this in my asterisk console: [Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18 (Ring Begin)... And indeed, Asterisk seems to take longer to pass the call to the destination extensions, exactly as if it's struggling to get the CID on the analog line. Does that message from Asterisk mean anything to anyone? b. signature.asc Description: This is a digitally signed message part ___ -- 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] zap not getting callerid any more
Brian J. Murrell wrote: One thing I have noticed is that in the cases where the wildcard cannot determine the CID (i.e. because the rxgain is up around 10.5), I get this in my asterisk console: [Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18 (Ring Begin)... And indeed, Asterisk seems to take longer to pass the call to the destination extensions, exactly as if it's struggling to get the CID on the analog line. Does that message from Asterisk mean anything to anyone? It means that Asterisk has detected that the line is ringing. The fact that Asterisk pauses after this indicates that it is waiting to receive caller ID information. The chances are that in boosting the receiving audio, you're causing the caller ID information to become distorted - enough so that Zaptel can no longer decode the caller ID properly. X100 cards are notorious for problems. Unless you want to invest in a better card, you may just have to live with the problem. ___ -- 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] Interfacing pri card to legacy pbx
Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom ___ -- 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] Incoming calls on zaptel not answered.
On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote: After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. THis isn;t going to fix your problem... but just FYI, you don't need to install libpri if you are just using a TDM400P (since its not a PRI / BRI [1.6 libpri does BRI as well] card). Might save you a little bit of time in the future, and its one less thing to consider as a problem. -- Matt Watson http://www.mattgwatson.ca ___ -- 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] Zaptel problem with pots lines
Hi Tzafrir, you're right. I think I've completely misunderstood the problem. If the problem is that asterisk is not able to write in the CDR the proper line answer status, I can confirm that even my installations behave the same. Sorry Enrico for my fault and thank you to Tzafrir for the correction. Best regards, Marco Signorini. Tzafrir Cohen wrote: On Mon, Jul 14, 2008 at 08:56:40PM +0200, Enrico Maistro wrote: Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. AFAIK chan_zap can only detect answer if it is provided through a polarity reversal. ___ -- 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] Interfacing pri card to legacy pbx
I cannot tell for sure for any system, but we have an old Portmaster PM3 hooked-up from one port of our Sangoma A104d card, another one being from telco. So, yes you can emulate the telco from a sangoma A10x card. Here's what I have in my zapata.conf : ;Sangoma A104 port 1 [slot:12 bus:0 span: 1] switchtype=national pridialplan=unknown signalling=pri_cpe group=1 channel = 1-23 ;Sangoma A104 port 2 [slot:12 bus:0 span: 2] echocancel=no pridialplan=national signalling=pri_net group=2 channel = 25-47 You might have noticed that the signalling is different for both port. pri_net being the telco emulatin one. The clock needs also to be set on master in the wancfg utility. Another thing, you might want to consider using a 2 port card for that, because the clock master needs a reference and I can't tell for sure if it'll work with a reference from another card. Regards, Nicolas Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom ___ -- 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] zap not getting callerid any more
On Tue, 2008-07-15 at 22:31 +1000, Rob Hillis wrote: Brian J. Murrell wrote: One thing I have noticed is that in the cases where the wildcard cannot determine the CID (i.e. because the rxgain is up around 10.5), I get this in my asterisk console: [Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18 (Ring Begin)... And indeed, Asterisk seems to take longer to pass the call to the destination extensions, exactly as if it's struggling to get the CID on the analog line. Does that message from Asterisk mean anything to anyone? It means that Asterisk has detected that the line is ringing. But I don't get that message when the rxgain is low enough for CID to work, yet the line is still picked up by Asterisk, so Asterisk must still be detecting the line ringing. Why does it print that message only when the rxgain is increased? The fact that Asterisk pauses after this indicates that it is waiting to receive caller ID information. Indeed. The chances are that in boosting the receiving audio, you're causing the caller ID information to become distorted - enough so that Zaptel can no longer decode the caller ID properly. Fair enough, but I'm only increasing to the magical values that are supposed to be ideal for echo cancellation. That seems to be a) incompatible with caller-id and b) it's only when I increase it that high do I get the Got event 18 (Ring Begin)... message. Further to (a) above, my rxgain has been at 10.9 for a long time and everything worked just fine and then one day the CID just stopped working. And further to that, a re-calibration to a milliwatt number showed it was only out by 4 points. Not very much it seems. X100 cards are notorious for problems. Supposedly, yet. But mine has been working peachy up until this out of the blue incident. Unless you want to invest in a better card, you may just have to live with the problem. Which means what, a multiport and multi-hundreds of dollar card? I'm just a home user. I don't have hundreds of dollars to spend on a single piece of phone hardware. I wonder how much using something like an SPA-3102 (with both an FXS and FXO ports) eliminates these problems and brings reliability to the table. b. signature.asc Description: This is a digitally signed message part ___ -- 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] Interfacing pri card to legacy pbx
Actually what I'm doing is interfacing the legacy pbx and converting it to use sip for its way out to the world. The phone vender I'm working with says his system requires b8zs signaling and uses the esf frame type. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Ross Sent: Tuesday, July 15, 2008 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Interfacing pri card to legacy pbx I cannot tell for sure for any system, but we have an old Portmaster PM3 hooked-up from one port of our Sangoma A104d card, another one being from telco. So, yes you can emulate the telco from a sangoma A10x card. Here's what I have in my zapata.conf : ;Sangoma A104 port 1 [slot:12 bus:0 span: 1] switchtype=national pridialplan=unknown signalling=pri_cpe group=1 channel = 1-23 ;Sangoma A104 port 2 [slot:12 bus:0 span: 2] echocancel=no pridialplan=national signalling=pri_net group=2 channel = 25-47 You might have noticed that the signalling is different for both port. pri_net being the telco emulatin one. The clock needs also to be set on master in the wancfg utility. Another thing, you might want to consider using a 2 port card for that, because the clock master needs a reference and I can't tell for sure if it'll work with a reference from another card. Regards, Nicolas Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom ___ -- 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] Interfacing pri card to legacy pbx
The configuration for a PM3 would be the same for a PBX. One additional note, put the channels on the PBX PRI in its own context, and then set that context up in your dialplan to forward the calls out to your SIP provider. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tom Moore Sent: Tuesday, July 15, 2008 9:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Interfacing pri card to legacy pbx Actually what I'm doing is interfacing the legacy pbx and converting it to use sip for its way out to the world. The phone vender I'm working with says his system requires b8zs signaling and uses the esf frame type. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Ross Sent: Tuesday, July 15, 2008 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Interfacing pri card to legacy pbx I cannot tell for sure for any system, but we have an old Portmaster PM3 hooked-up from one port of our Sangoma A104d card, another one being from telco. So, yes you can emulate the telco from a sangoma A10x card. Here's what I have in my zapata.conf : ;Sangoma A104 port 1 [slot:12 bus:0 span: 1] switchtype=national pridialplan=unknown signalling=pri_cpe group=1 channel = 1-23 ;Sangoma A104 port 2 [slot:12 bus:0 span: 2] echocancel=no pridialplan=national signalling=pri_net group=2 channel = 25-47 You might have noticed that the signalling is different for both port. pri_net being the telco emulatin one. The clock needs also to be set on master in the wancfg utility. Another thing, you might want to consider using a 2 port card for that, because the clock master needs a reference and I can't tell for sure if it'll work with a reference from another card. Regards, Nicolas Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom ___ -- 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
[asterisk-users] Reinvites and SIP/RTP
Hi All, When I use re-invite, does the Asterisk server stay in the SIP conversation, and just RTP traffic diverts, or does the SIP transfer away from the A*k server too ? Thanks, Adrian ___ -- 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] Music on hold
Hi, I'm getting this bizarre problem. Whenever I dial (through misdn) and try to listen to my music on hold, I get this: -- Started music on hold, class 'default', on channel 'mISDN/3-u72' [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 8192 requested bytes on mISDN/3-u72 [Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate: Only doing 2624 of 6644 requested bytes on mISDN/3-u72 -- Stopped music on hold on mISDN/3-u72 Any idea??? Thanks :D ___ -- 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] US T1 Hangup Detection
On Fri, Jul 11, 2008 at 03:59:22PM -0500, Joe Greco wrote:
On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote:
Really? You have an RJ-21X block that contains both analog AND T1
wires? That's really uncommon. I hope they at least put the red
special service caps on the T1 wires.
Yup. I thought that pretty funny myself. 10 year old analog wires
running a digital T1. :) And they do have some caps on them, I think
it was red but not 100% sure.
No, that's not the unusual part. The unusual part is just that both
analog and digital services are on the same block. Maybe it's a
regional think...
That's really not unusual. It's not /preferred/, but that's an entirely
different can of worms.
I'll bet. :-)
In general, if copper is available into a building, the telco is going to
look very seriously at the possibility of using that. If the building is
already wired and the copper tests clean, the telco will want to use that.
In most existing situations, that will already be terminated in a can with
lightning suppression and will have been crossed over to RJ21X's that are
going to whatever suites are in the building.
So we don't pay a lot of attention to Tx and Rx in separate jackets,
or shielded anymore? Or is so much T-1 delivery over 1-pair HDSL that
no one cares anymore?
Since the telco will have /no/ /problem/ running the T1 over their outside
plant and up to the can on what is approximately Category 3 wire, and the
T1 signal is going to have been running alongside those same analog wires
for probably a few miles, what happens next should be obvious.
Cat 3 is optimistic, IME. Cat 2 is good enough for T-1, though; I
looked once.
Suite 214 wants a T1. There's already a 25-pair going up there from the
RJ21X. It's second story, so do you go and spend an {hour, afternoon,
etc} figuring out how to run fresh wire, or do you notice that only 6 pair
are in use on the RJ21X, and decide to feed up on the existing cable?
Now, if you're nasty and you don't separate it (typically I see the bottom
used for data) and you don't put redcaps on, yeah, then that is just
looking for eventual trouble. And who knows, the wire may be cruddy, so
maybe you still end up doing the separate run. But it probably works.
I've seen this often enough. Would I prefer to see new cable run? Sure.
But we've all done our copper sins. I've seen a lot of things that are
uglier than that. Here's one of them:
http://www.sol.net/hallofshame/
Slithering jesus. :-)
(I've always meant to expand that page, but it seems that I never get the
good photos of bad stuff)
I was going to ask...
Lack of space, lack of need, lack of having another RJ21X in the truck are
just a few other obvious reasons that this might be done.
True.
Your netmon link is 404, BTW.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
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Re: [asterisk-users] Music on hold
Vazquez David wrote:
Hi,
I'm getting this bizarre problem. Whenever I dial (through misdn) and
try to listen to my music on hold, I get this:
-- Started music on hold, class 'default', on channel 'mISDN/3-u72'
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 8192 requested bytes on mISDN/3-u72
[Jul 15 17:15:15] WARNING[13393]: res_musiconhold.c:742 moh_generate:
Only doing 2624 of 6644 requested bytes on mISDN/3-u72
-- Stopped music on hold on mISDN/3-u72
Any idea???
Thanks :D
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Solved :
I didn't have an answer statement in my extensions.conf
The working context:
exten = 03,1,Answer()
exten = 03,2,Queue(${EXTEN})
exten = 03,3,Hangup()
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[asterisk-users] sip prune realtime per issue
I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in sip show peer , but everything is not being updated. The phone will not register even though the DB and the phone have the correct password. The only way to get it to register is to stop * and re-start it, then it works fine. I even tried changing the callerid and pruned the peer. A sip show peer shows the correct callerid, but when you call into voicemail, it is using the old callerid. Again, if I stop * and restart, it works fine. Has anybody seen this bug and if so, know what the bug ID is? We have a bunch of patches on these boxes and can't just upgrade to any old version to see if it fixes it. I need to figure out what the bug is. I did some research, but couldn't find it. Peder ___ -- 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] zap not getting callerid any more
One thing I have noticed is that in the cases where the wildcard cannot determine the CID (i.e. because the rxgain is up around 10.5), I get this in my asterisk console: [Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18 (Ring Begin)... It is odd that it would work one day and not the next. I'd have to say, though that I've seen that rxgain/txgain values beyond +-8 seem to yield unpredictable results in many areas, even if they do get you closer to 14844, and that's even on the cool new cards all the kids are using these days. And now the obligatory: YMMV - Noah ___ -- 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] Reinvites and SIP/RTP
Hi Adrian - When I use re-invite, does the Asterisk server stay in the SIP conversation, and just RTP traffic diverts, or does the SIP transfer away from the A*k server too ? I'm sure somebody will correct me if this is wrong, but I believe the signalling must stay with asterisk, as asterisk needs to know if it should provide any services for the call (music on hold, transfer, etc). - Noah ___ -- 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] sip prune realtime per issue
On Tue, Jul 15, 2008 at 12:05 PM, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in sip show peer , but everything is not being updated. The phone will not register even though the DB and the phone have the correct password. The only way to get it to register is to stop * and re-start it, then it works fine. I even tried changing the callerid and pruned the peer. A sip show peer shows the correct callerid, but when you call into voicemail, it is using the old callerid. Again, if I stop * and restart, it works fine. Has anybody seen this bug and if so, know what the bug ID is? We have a bunch of patches on these boxes and can't just upgrade to any old version to see if it fixes it. I need to figure out what the bug is. I did some research, but couldn't find it. Peder Do the rt* options in sip.conf have any effect? Maybe one of those might help? --Marc ___ -- 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 !DSPAM:1,487ccb5365666785646901! ___ -- 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] zap not getting callerid any more
On Tue, 2008-07-15 at 12:49 -0400, Noah Miller wrote: It is odd that it would work one day and not the next. Indeed. I'd have to say, though that I've seen that rxgain/txgain values beyond +-8 seem to yield unpredictable results in many areas, Yeah, I was pretty alarmed months ago when I tuned it to 10.9 to get that magical 14844, but there was no echo and CID worked, but given everything was working, didn't really concern myself with it further. even if they do get you closer to 14844, and that's even on the cool new cards all the kids are using these days. And now the obligatory: YMMV Indeed. Things seem to be working again with rxgain at 10.0. Any attempt to push it closer to that 10.5 (from today's calibration) or 10.9 (calibration of a few months ago) yield CID problems again. But I still wonder why the higher values result in chan_zap.c:6670 ss_thread: Got event 18 (Ring Begin)... messages and lower values do not AND YET Asterisk answers the Zap/POTS line in either case. I can't help but think that message and the lack of CID at the higher, calibrated value are related. b. signature.asc Description: This is a digitally signed message part ___ -- 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] distintive ring
Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? ___ -- 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] Toll Free International Number
Hello All, I am looking to find a way to provide international toll free access to our Knoxville, TN (USA) office from our customers in the UK and in Australia, and when I talked with ATT I was surprised to find out how expensive they are... Surely, other businesses are not paying this much - are they?!?! Can someone in this good group please help me with some advice as to who can provide affordable and reliable international toll free service for a better price than ATT? Thanks in advance, Larry Costigan Food Donation Connection (Asterisk fan and ABE user) ___ -- 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] Toll Free International Number
Larry, Give us a call (646) 862-1555 /jon - Original Message - From: Larry Costigan To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, July 15, 2008 2:22 PM Subject: [asterisk-users] Toll Free International Number Hello All, I am looking to find a way to provide international toll free access to our Knoxville, TN (USA) office from our customers in the UK and in Australia, and when I talked with ATT I was surprised to find out how expensive they are... Surely, other businesses are not paying this much - are they?!?! Can someone in this good group please help me with some advice as to who can provide affordable and reliable international toll free service for a better price than ATT? Thanks in advance, Larry Costigan Food Donation Connection (Asterisk fan and ABE user) -- ___ -- 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
[asterisk-users] Meetme replacement with native 729 support
Folks: Does anyone know of a replacement for meetme that provides native G729 support? The transcoding back and forth from/to 711 is eating too much processor for what we're doing... Many thanks, --ag -- Artie Gold F4W, Inc. ___ -- 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] distintive ring
Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? Fidel, I do not know what kind of tone you mean: The sound of a phone that signals a call coming from internal/external? The sound in the earpiece after you dialled while you wait for the other end to pick up? In the first case distinctive ring is probably the right term to search for. You will have to decide wether your phones are SIP or ZAP (or both, or different), because methods seem to differ. As a start reading point have a look at http://www.malico.com.tw/voip-info/wiki/view/Asterisk+SIP+channels.html The mailing list archives contain a lot of information *hint* Best regards Anselm ___ -- 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] distintive ring
This one! The sound of a phone that signals a call coming from internal/external My phones are SIP, I do not know what ZAP means or what it does. Thanks for your reply! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 15, 2008 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] distintive ring Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? Fidel, I do not know what kind of tone you mean: The sound of a phone that signals a call coming from internal/external? The sound in the earpiece after you dialled while you wait for the other end to pick up? In the first case distinctive ring is probably the right term to search for. You will have to decide wether your phones are SIP or ZAP (or both, or different), because methods seem to differ. As a start reading point have a look at http://www.malico.com.tw/voip-info/wiki/view/Asterisk+SIP+channels.html The mailing list archives contain a lot of information *hint* Best regards Anselm ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.4.11/1553 - Release Date: 7/15/2008 5:48 AM ___ -- 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] distintive ring
Internal and external calls can be distinguished generally by the phone number. A prefix or the number of digits of the phone number. For example, you could use a digit prefix followed by a interval of time to call a internal number. Examples: Internal number: 0,1234 External number: 87654321 On Tue, Jul 15, 2008 at 2:02 PM, Fidel Garcia [EMAIL PROTECTED] wrote: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net -- ___ Allann J. O. Silva I received the fundamentals of my education in school, but that was not enough. My real education, the superstructure, the details, the true architecture, I got out of the public library. For an impoverished child whose family could not afford to buy books, the library was the open door to wonder and achievement, and I can never be sufficiently grateful that I had the wit to charge through that door and make the most of it. (from I. Asimov, 1994) ___ -- 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] distintive ring
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels On Tue, Jul 15, 2008 at 2:37 PM, Fidel Garcia [EMAIL PROTECTED] wrote: This one! The sound of a phone that signals a call coming from internal/external My phones are SIP, I do not know what ZAP means or what it does. Thanks for your reply! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 15, 2008 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] distintive ring Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? Fidel, I do not know what kind of tone you mean: The sound of a phone that signals a call coming from internal/external? The sound in the earpiece after you dialled while you wait for the other end to pick up? In the first case distinctive ring is probably the right term to search for. You will have to decide wether your phones are SIP or ZAP (or both, or different), because methods seem to differ. As a start reading point have a look at http://www.malico.com.tw/voip-info/wiki/view/Asterisk+SIP+channels.html The mailing list archives contain a lot of information *hint* Best regards Anselm ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.4.11/1553 - Release Date: 7/15/2008 5:48 AM ___ -- 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 -- ___ Allann J. O. Silva I received the fundamentals of my education in school, but that was not enough. My real education, the superstructure, the details, the true architecture, I got out of the public library. For an impoverished child whose family could not afford to buy books, the library was the open door to wonder and achievement, and I can never be sufficiently grateful that I had the wit to charge through that door and make the most of it. (from I. Asimov, 1994) ___ -- 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] Incoming calls on zaptel not answered.
Hi, I need libpri, because I have a TE110P E1 with a PRI ISDN service. 2008/7/15 Matt Watson [EMAIL PROTECTED]: On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote: After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. THis isn;t going to fix your problem... but just FYI, you don't need to install libpri if you are just using a TDM400P (since its not a PRI / BRI [1.6 libpri does BRI as well] card). Might save you a little bit of time in the future, and its one less thing to consider as a problem. -- Matt Watson http://www.mattgwatson.ca ___ -- 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 -- Jose Flores Galicia [EMAIL PROTECTED] BriefCode Code Based Training ___ -- 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] Incoming calls on zaptel not answered.
Thank you, yes, I changed the PCI Slot and it's the same, I get a used card from a customer with 2 FXO, same REV, that board was working on the customer server, put it on mine, and stop working. I put my board on his server and the board is working perfectly. I had not test outgoing calls on that board, I tried and outgoing works fine. 2008/7/15 Noah Miller [EMAIL PROTECTED]: Hi Jose - After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. The board is working, I tested in another server with the 1.2.13 asterisk version. Also changed the pci slot where the board is. Hmm. Bad or incompatible PCI slot? Can you (at least for testing purposes) switch back to the original PCI slot you were using when the card worked? - Noah ___ -- 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 -- Jose Flores Galicia [EMAIL PROTECTED] BriefCode Code Based Training ___ -- 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] Meetme replacement with native 729 support
On Tuesday 15 July 2008 13:32:12 Artie Gold wrote: Does anyone know of a replacement for meetme that provides native G729 support? The transcoding back and forth from/to 711 is eating too much processor for what we're doing... Buy a hardware transcoder board. There is simply no way to mix compressed audio like that without decompressing first. And by the way, it's decompressing to signed linear 16-bit audio, not ulaw. Even mixing of ulaw requires a decompress to signed linear. -- Tilghman ___ -- 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] Meetme replacement with native 729 support
That makes sense -- thanks! --ag On Tue, Jul 15, 2008 at 1:59 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 15 July 2008 13:32:12 Artie Gold wrote: Does anyone know of a replacement for meetme that provides native G729 support? The transcoding back and forth from/to 711 is eating too much processor for what we're doing... Buy a hardware transcoder board. There is simply no way to mix compressed audio like that without decompressing first. And by the way, it's decompressing to signed linear 16-bit audio, not ulaw. Even mixing of ulaw requires a decompress to signed linear. -- Tilghman ___ -- 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 -- Artie Gold F4W, Inc. ___ -- 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] US T1 Hangup Detection (Resolved)
On Jul 11, 2008, at 12:58 PM, Daniel Hazelbaker wrote: I may have figured out the problem this morning, but I won't be able to test for a few days (again, aggravating that the only T1 line I have to test with is the live one). I noticed this morning while telneted into the Adtran that when I hangup on our normal incoming lines the Receive A bit toggles. I then noticed that two of the lines do NOT toggle the RA bit during hangup. These happen to the be last two lines in the rotary so I would not normally get incoming calls and complaints on them. They also happen to be the lines I was using to do my testing with. Grrr. Just to close out this thread for anybody interested, last night I hooked up the T1 line again and verified that this was indeed the problem. Out of the 12 lines in use on the T1, 4 of them do not provide the disconnect supervision. So I have called and updated my trouble ticket to include all 4 of those channels. Thanks again everybody for the suggestions and bits of information that helped me track down this problem. Daniel ___ -- 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] Meetme replacement with native 729 support
OK, I guess I need to show my ignorance -- what is the difference between ulaw and signed linear? on Tuesday 07/15/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote On Tuesday 15 July 2008 13:32:12 Artie Gold wrote: Does anyone know of a replacement for meetme that provides native G729 support? The transcoding back and forth from/to 711 is eating too much processor for what we're doing... Buy a hardware transcoder board. There is simply no way to mix compressed audio like that without decompressing first. And by the way, it's decompressing to signed linear 16-bit audio, not ulaw. Even mixing of ulaw requires a decompress to signed linear. -- Tilghman ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- 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] Toll Free International Number
Depending upon what cities you need, there are a lot of companies offering this. I like IdeaSIP.com who have shown excellent call quality and value over the years I've been using them. /r Can someone in this good group please help me with some advice as to who can provide affordable and reliable international toll free service for a better price than ATT? ___ -- 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 not receive calls through pri
Hi Uros - I have problem using Asterisk.I have isdn-pri and openvox d110p card in my computer.They are connected with RJ-45 (1,2,4,5 pins to the card and all pins to the isdn done by telco workers). I got green led on isdn which is sign that isdn is working and that is connected to openvox, right ? I compiled newest versions of libpri zaptel and asterisk and had no problems during that. When I started services I can not receive any calls.No indication that any call is coming to Asterisk.When I dial number (to my line coz it is IN service so they can only call me not other way) I can hear telco message then few seconds of silence and busy signal. On cli I can not see anything.By the way I use Fedora 9 x64 kernel (I tryed with i386 kernel, with different machines,different distributions too but same problem occurred. Just to double-check: did you use the patched wcte11xp.c file from the openvox website? - Noah ___ -- 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] distinctive ring
It depends on which type of SIP device you have that determines on how you signal a distinctive ring. You need to change the SIP Header like: exten = s,n,SIPAddHeader(Alert-Info:Bellcore-r8) where the number after the 'r' signifies a different ring tone but some devices uses different names other than Bellcore... If you are on an internal path you would set one ring and if you are on an external path set another. Fidel Garcia wrote: This one! The sound of a phone that signals a call coming from internal/external My phones are SIP, I do not know what ZAP means or what it does. Thanks for your reply! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 15, 2008 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] distintive ring Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? Fidel, I do not know what kind of tone you mean: The sound of a phone that signals a call coming from internal/external? The sound in the earpiece after you dialled while you wait for the other end to pick up? In the first case distinctive ring is probably the right term to search for. You will have to decide wether your phones are SIP or ZAP (or both, or different), because methods seem to differ. As a start reading point have a look at http://www.malico.com.tw/voip-info/wiki/view/Asterisk+SIP+channels.html The mailing list archives contain a lot of information *hint* Best regards Anselm ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.4.11/1553 - Release Date: 7/15/2008 5:48 AM ___ -- 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
[asterisk-users] How to monitor Asterisk logs ?
Hi, How can I be notified anytime a given warning message appears in Asterisk logs ? I've got a running system that produces cause 34 warnings (Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)) once or twice a week. I would like to like to be notified (by email, phone, ...) anytime such warning message occurs in log file. I was thinking of using logwatch but wondered if anything better exists. Any advice ? Regards ___ -- 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] Meetme replacement with native 729 support
On Tuesday 15 July 2008 14:24:30 John covici wrote: on Tuesday 07/15/2008 Tilghman Lesher([EMAIL PROTECTED]) wrote On Tuesday 15 July 2008 13:32:12 Artie Gold wrote: Does anyone know of a replacement for meetme that provides native G729 support? The transcoding back and forth from/to 711 is eating too much processor for what we're doing... Buy a hardware transcoder board. There is simply no way to mix compressed audio like that without decompressing first. And by the way, it's decompressing to signed linear 16-bit audio, not ulaw. Even mixing of ulaw requires a decompress to signed linear. OK, I guess I need to show my ignorance -- what is the difference between ulaw and signed linear? ulaw is a compression algorithm which compresses the 16-bit 8000Hz signed linear (slin) format down to 8-bits per sample. So while signed linear consumes 128kbps, ulaw only consumes 64kbps. Ulaw is actually a fairly simple coding algorithm, and it compresses from and decompresses to slin with a 1-to-many lookup table between the values. So it's pretty fast, as implemented. A more technical explanation can be found on Wikipedia, if you are so inclined: http://en.wikipedia.org/wiki/Μ-law_algorithm -- Tilghman ___ -- 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] ASTERISK/ENSWITCH ON EC2
On Jul 11, 2008, at 12:28 PM, Robert McNaught wrote: Has anyone deployed a hosted environment like enswitch using EC2? I was wondering if anyone had any thoughts on concerns on the feasibility in doing this using cloud computing? For setting up a VoIP service provider and not having the headache of dealing with the hassle and expenses of hardware, racks, cages etc, it looks pretty attractive. Any thoughts? If you are setting up a VoIP service provider, I would be concerned about the service uptime using the EC2 cluster, it is after all still in beta and had a number of outages over the past year. Have you considered other hosting solutions? There are a number of high quality hosting offerings, that offer 24/7 phone support, without requiring any long term contracts. -- Eric Chamberlain Founder RF.com http://RF.com/ ___ -- 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] (no subject)
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I try to build Asterisk this is the error I'm getting. src/add.c:1: error: CPU you selected does not support x86-64 instruction set I just can't seem to find what i need to set to get this to build. Thanks _ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_WL_Refresh_messenger_video_072008___ -- 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] Adtran IP712
All: Has anyone else on the list had any experience with the new Adtran IP712 phones? I have taken the stock config file and been able to get simple registrations and basic call processing to work properly, however, I'm finding little to no documentation on how to configure advanced options such as BLF/Parking/Definitions for Files (Phonebook,xml,etc). I guess I may be assuming a lot about the phones, but I was hoping for this type of functionality as these phones are listed as Digium|Asterisk Premier Interoperability Partner. Overall, I am happy with the phones themselves, but I'm looking for anyone else with experience configuring features to work with Asterisk 1.4.19. If you have any information or suggestions I'd appreciate it! Thanks, Joshua Tressler Network Engineer Enhanced Telecommunications Corporation Office: (812) 222-1020 Cell: (812) 593-0314 Email: [EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ -- 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] How to monitor Asterisk logs ?
perl script. Olivier wrote: Hi, How can I be notified anytime a given warning message appears in Asterisk logs ? I've got a running system that produces cause 34 warnings (Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)) once or twice a week. I would like to like to be notified (by email, phone, ...) anytime such warning message occurs in log file. I was thinking of using logwatch but wondered if anything better exists. Any advice ? Regards ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- 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] (no subject)
Hi - I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I try to build Asterisk this is the error I'm getting. src/add.c:1: error: CPU you selected does not support x86-64 instruction set You may not have the right sources for your kernel. You may have the 32-bit sources instead of the 64-bit ones. What kind of CPU is it? - Noah ___ -- 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] gui issue in asterisk aa50
HI all, I am having issues with the gui on my AA50. under Voice Menus Add new Step Go to Time based rule. It allows me to select Go to Time based rule from the menu but no options come up when selected. I've tried all browsers but no luck. Thanks 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
Re: [asterisk-users] Toll Free International Number
voxbone.com - Original Message - From: Larry Costigan To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Toll Free International Number Date: Tue, 15 Jul 2008 14:22:49 -0400 Hello All, I am looking to find a way to provide international toll free access to our Knoxville, TN (USA) office from our customers in the UK and in Australia, and when I talked with ATT I was surprised to find out how expensive they are... Surely, other businesses are not paying this much - are they?!?! Can someone in this good group please help me with some advice as to who can provide affordable and reliable international toll free service for a better price than ATT? Thanks in advance,Larry CostiganFood Donation Connection (Asterisk fan and ABE user) ___ -- 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 -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com ___ -- 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] Beginner Issues
I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- 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] Beginner Issues
Hi John - I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? Most probably it's a codec issue, but we'll need to see your sip.conf file. It might also be helpful to know what SIP devices you're using. - Noah ___ -- 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] Beginner Issues
Are your phones behind NAT? This should be an issue with rtp port communication. Gerard. --Original Message-- From: John Koenig Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Jul 15, 2008 6:47 PM Subject: [asterisk-users] Beginner Issues I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- 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 Sent from my T-Mobile BlackBerry Handheld ___ -- 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] distintive ring
My internal calls start in an entirely different context than calls coming in externally. There's never any confusion about where the call is coming from and I don't use prefixes. Allann Jones wrote: Internal and external calls can be distinguished generally by the phone number. A prefix or the number of digits of the phone number. For example, you could use a digit prefix followed by a interval of time to call a internal number. Examples: Internal number: 0,1234 External number: 87654321 On Tue, Jul 15, 2008 at 2:02 PM, Fidel Garcia [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net -- ___ Allann J. O. Silva I received the fundamentals of my education in school, but that was not enough. My real education, the superstructure, the details, the true architecture, I got out of the public library. For an impoverished child whose family could not afford to buy books, the library was the open door to wonder and achievement, and I can never be sufficiently grateful that I had the wit to charge through that door and make the most of it. (from I. Asimov, 1994) ___ -- 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] gui issue in asterisk aa50
Have you upgraded to the latest version? We found a few bugs went away on our test unit when we did that. PaulH Sydney Web Hosting wrote: HI all, I am having issues with the gui on my AA50. under Voice Menus Add new Step Go to Time based rule. It allows me to select “Go to Time based rule” from the menu but no options come up when selected. I’ve tried all browsers but no luck. Thanks 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 ___ -- 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] gui issue in asterisk aa50
I had an issue where I put a comma in the prepend digits string pn call plans and then the call plan menu would no longer load. It parses the menu from the text file so I used the file editor to clear the offending line and my menu came back. Not sure if thats your issue but I was surprised I could enter text that broke the menus Cheers Duncan On 16/07/2008, at 10:27 AM, Sydney Web Hosting [EMAIL PROTECTED] wrote: HI all, I am having issues with the gui on my AA50. under Voice Menus Add new Step Go to Time based rule. It allows me to select “Go to Time based rule” from the menu but no options come up when selected. I’ve tried all browsers but no luck. Thanks 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 ___ -- 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] Beginner Issues
I had issues like this on one installation that cleared up when I turned ACPI and APIC?? off in bios. Darren Wiebe [EMAIL PROTECTED] Gerard A. Matthew wrote: Are your phones behind NAT? This should be an issue with rtp port communication. Gerard. --Original Message-- From: John Koenig Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Jul 15, 2008 6:47 PM Subject: [asterisk-users] Beginner Issues I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- 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 Sent from my T-Mobile BlackBerry Handheld ___ -- 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] Beginner Issues
That could be...I only have ports 5060 and 8088 open on the firewall. Should another port be open? The phone I am using are pstn phones connected to a 2 port Linksys PAP2. I made sure that I checked the NAT option under the user account and enabled NAT Keep Alive under the PAP2 management interface. I am using the G726-16 codec for transmission. Attached is my sip.conf. John Gerard A. Matthew wrote: Are your phones behind NAT? This should be an issue with rtp port communication. Gerard. --Original Message-- From: John Koenig Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Jul 15, 2008 6:47 PM Subject: [asterisk-users] Beginner Issues I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- 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 Sent from my T-Mobile BlackBerry Handheld ___ -- 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 ; ; SIP Configuration example for Asterisk ; ; Syntax for specifying a SIP device in extensions.conf is ; SIP/devicename where devicename is defined in a section below. ; ; You may also use ; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet ; (Don't forget to enable DNS SRV records if you want to use this) ; ; If you define a SIP proxy as a peer below, you may call ; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED] ; where the proxyhostname is defined in a section below ; ; Useful CLI commands to check peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show users Show all SIP users (including friends) ; sip show registry Show status of hosts we register with ; ; sip debug Show all SIP messages ; ; reload chan_sip.so Reload configuration file ; Active SIP peers will not be reconfigured ; [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) ; Default is enabled ;realm=mydomain.tld ; Realm for digest authentication ; defaults to asterisk. If you set a system name in ; asterisk.conf, it defaults to that system name ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) ; bindport is the local UDP port that Asterisk will listen on bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ;domain=mydomain.tld; Set default domain for this host ; If configured, Asterisk will only allow ; INVITE and REFER to non-local domains ; Use sip show domains to list local domains ;pedantic=yes ; Enable checking of tags in headers,
Re: [asterisk-users] gui issue in asterisk aa50
Time based rules are no longer in use. Contact Digium support that you got received with your aa50. -bk ___ -- 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] changing inbuilt sound messages
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Lists wrote: Hi all, I am wanting to change the sound files from the standard ones to a New Zealand voice pack. I have copied the files into the /var/lib/asterisk/sounds directory and chowned them to asterisk:asterisk and chmod 420 to match the existing files but the system is still using the original files. The original files seem to be wav files while the NZ voice pack ones are gsm files. How do I get the system to use the new gsm files? Bear in mind that the New Zealand sound files are only available in GSM. Asterisk will choose the sound file which best matches the current audio format. I.E. if you have ulaw/alaw sounds and a ulaw/alaw conversation, Asterisk will use the alaw/ulaw sounds rather than the GSM ones. Either remove all files and only install GSM (and then put NZ over the top), or convert the NZ sounds to ulaw/alaw. Alternatively, make sure your call is in GSM. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfU5oDQNt8rg0Kp4RAtBfAJ0dZVK9J8Qki5B01ZXMX6oiKqf7VgCeLVnA MpO+/VOYVYvQ7Ckz3JCfMZo= =REJx -END PGP SIGNATURE- ___ -- 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] RTP packets dropped
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Vinícius Fontes wrote: As RTP packets have a sequential number, is there some logging/debugging option in Asterisk to monitor how many packets have been lost on a SIP call? You could use rtcp stats if the endpoints support it. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfU7PDQNt8rg0Kp4RAo1dAKCNUKO3NvVKnce7FNk2rI/4D1YfQwCfSXMl T+0EYmctykhpP3he1FCQiPY= =EZSI -END PGP SIGNATURE- ___ -- 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] Diagnosing dropped calls...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John Faubion wrote: Try dropping the IAX2 and only use SIP. Don't ask why? Well in our case we were NOT using IAX at all. Strictly SIP. You could be hitting an overloaded router or whatever along the way, 10mbs fiber does not mean low latency or lost packets. So true, hence the reason I suggested using mtr to check it. Many times in our case we saw gateways between networks that were dropping packets presumably due to overload conditions. RTP traffic over UDP would add far more load than the ICMP packets used for mtr. Yes and no. I've seen pretty major problems show up in mtr only to be told that the provider is dropping icmp in times of high load. We moved our monitoring to sip/iax times, but you only get a point to point stat in that situation. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVB6DQNt8rg0Kp4RArC0AKCHRQd3RCenmNDN/E4M3+pCDqTxuACgsjFs Lk8f3IgqMF1uPU3RKemQKJg= =VU7t -END PGP SIGNATURE- ___ -- 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] Incoming
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Artie Gold wrote: Folks: This is my first post, so please let me know if I transgress in any way... In updating to 1.4.21 recently, we've encountered a problem, when running over a satellite connection (where the latency is considerable; a regular internet connection did not exhibit this problem), where incoming calls are being dropped as a result of the sip handshake timing out (dropping down to 1.4.18.1 solved the problem for us). From reading the change logs and other posts, it seems that some work has been done in this area recently to get it right; it appears that, at least in the satellite case, things may have gotten a little too tight... If this rings a bell for anyone, any insight would be appreciated. These calls sip or iax? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVFWDQNt8rg0Kp4RAvj3AJ0bKXhQDS5v8bqAOQF9llPZdTh/wQCfclx3 vXJlSU/zoQY4mUxQhKE3mTY= =oFsV -END PGP SIGNATURE- ___ -- 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] Sipura 3000 replacement --- SPA3102 how reliable is it?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Underwood wrote: Dave Cotton wrote: Joseph wrote: On 07/11/08 18:37, Dave Cotton wrote: SIP wrote: Joseph wrote: I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? We have a few dozen subscribers using them at any given point in time. I and my wife even use them at our respective homes. Rock solid stable. No issues whatsoever. The only reservation I've got with the 3000/3102 units is that I've had 3 destroyed by lightening recently. But I'm told it's because I'm on the end of 3kms of cable across open countryside. The others I've installed in non rural installations work faultlessly. DC If you plug it into to UPS some of them have protection for phone lines, it should protect it from lightning. Should is the operative word. They didn't. DC I'm very suspicious of the effectiveness of the things they put in low end UPSes. However, if you buy the kind of lightning suppressor that is attached to phone lines as the enter your house, and put one at each end of your 3km of cable, it should help a lot. The APC UPSes we use come with a warranty for equipment attached to them, as do the belkin filters. Check if yours does any you may be able to get the units replaced at no cost. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVG6DQNt8rg0Kp4RAuSTAJ91LkVnGPoCa+DGDMe8mxAqDvC91wCgqQB+ 4egBNtmFgigz+ECMS2v4pyI= =m3aR -END PGP SIGNATURE- ___ -- 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] Recharge Dial Limit....?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Thanks, but how does that extend the core functionality of Dial()? If Dial() can't do it, how does a wrapper do it? Did you see the patch that someone pointed out in your last conversation? That does exactly that. If you didn't want a patched system you could do it the same as I did (albeit a number of years ago) for call shops. When a person walks in they pay $5 (or whatever at the desk). You update the db and assign them a phone. A second process is connected to Asterisk via the manager for call control. It checks the DB every few seconds, and updates the credit based on how long the person has been talking and the rate to that destination. They can add more money at any time, and the DB value is updated. When they run out of credit the call is killed via the manager. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVTBDQNt8rg0Kp4RAjbLAKCGQjQCFP4dZT7GMjCSomNmUKHlKQCdH+EW Rf8imcVun0O2IMH47zwylfg= =nmoF -END PGP SIGNATURE- ___ -- 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] XORCOM BRI interfaces
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Loic Didelot wrote: Hello, I just got my Xorcom BRI bank. Seems to work. But I have some questions. Is anyone getting good values using zttest? Is it plugged into the BRI? Is it the sync master? i.e. xpp_sync - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVZ4DQNt8rg0Kp4RAt0hAJ4029j/sh/HG5bcktRFCzYTs5y8HgCdGvLx ehhwenzbGTA9Vr/n2rYNTvE= =InwQ -END PGP SIGNATURE- ___ -- 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] Poor audio quality with TDM400 card
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Noah Miller wrote: Hi Leotis - Now that you mention that, i didnt even know there was a gsm bug. I am using asterisk 1.4.21.1, i visited the link you gave. I am guessing i will have to patch my asterisk installation, i am reading, the bug report to see,how i can verify that i have the gsm bug. Well, if you have gcc version 4.2.x (you can check with gcc -v) there's a good chance this is the problem. Just do: export CC=gcc-4.1 export CXX=gcc-4.1 ./configure make Works for me. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVX8DQNt8rg0Kp4RAnXWAKCETjshfkGADZxQ+Ne1cy2dSfx/6ACeLFUo gjydjPdX6ke1Udp6rnq7GIY= =yTVS -END PGP SIGNATURE- ___ -- 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] asterisk + web services
List, We're working on an upcoming job that may require us to access a web service (WS). I'm curious to hear peoples thoughts on the best way to do this with asterisk. We'll be submitting a single number to the WS and it will return a success or error. One solution would be to write a simple perl script to interface into to the WS, and use SYSTEM() from asterisk to call it. Another may be to use the func_curl to do it too. If anybody have suggestions / ideas please post them. Thanks again, PB ___ -- 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] QueueMemberStatus
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jason Dixon wrote: On Tue, Jul 08, 2008 at 11:00:43AM -0400, Jason Dixon wrote: On Tue, Jul 08, 2008 at 12:10:05PM +1200, Matt Riddell wrote: Action: Command Command: show queue my_queue_name ActionID: my_queue_name_12345 This does not appear to show the correct status of an extension. It appears that ExtensionState also always reports Status of -1. Are there any Actions or Commands that will report the correct status of an extension? So far the only accurate representation I've found of queue members has been the following. $ sudo /usr/sbin/asterisk -r -x show channels | grep '^SIP' SIP/241-b742e010 [EMAIL PROTECTED]:2Ring Dial(Zap/G1/411) $ sudo /usr/sbin/asterisk -r -x show queue support_queue | grep SIP SIP/207 (Ringing) has taken no calls yet SIP/203 (Not in use) has taken no calls yet SIP/202 (In use) has taken no calls yet SIP/201 (Not in use) has taken no calls yet All of the commands I've tried via the AGI have yielded incorrect results. If this sounds wrong, please let me know and I'll resume beating my head against the nearest wall. :) Looks correct. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVfuDQNt8rg0Kp4RAhsdAKCDJ6ya2YJPpSBxPemi88mV9sUf5gCfSNg0 i/cHg7W3yYGL8apTWBejPts= =6pTt -END PGP SIGNATURE- ___ -- 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] asterisk + web services
On Jul 15, 2008, at 10:08 PM, Paul Belanger wrote: List, We're working on an upcoming job that may require us to access a web service (WS). I'm curious to hear peoples thoughts on the best way to do this with asterisk. We'll be submitting a single number to the WS and it will return a success or error. Honestly, if you're running a webservice, I like using the CURL function. Works like a charm for me. Fred Posner [EMAIL PROTECTED] Tel: +1 (212) 937-7844 x501 smime.p7s Description: S/MIME cryptographic signature ___ -- 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] asterisk + web services
Try Adhersion and or Telegraph -E http://mobiquity.ws ___ -- 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] asterisk + web services
On Tue, 15 Jul 2008, Paul Belanger wrote: We're working on an upcoming job that may require us to access a web service (WS). I'm curious to hear peoples thoughts on the best way to do this with asterisk. We'll be submitting a single number to the WS and it will return a success or error. One solution would be to write a simple perl script to interface into to the WS, and use SYSTEM() from asterisk to call it. Another may be to use the func_curl to do it too. If anybody have suggestions / ideas please post them. curl() doesn't fire up another process. The response is returned as just one big chunk. In my case, it was the HTML to an entire web page :) If you need to do a bunch of parsing, maybe an AGI calling libcurl -- saving a bunch of ugly dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] (announce) asterisk T.38 gateway
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Totaro wrote: On Thu, Jul 10, 2008 at 11:43 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood [EMAIL PROTECTED] wrote: Vinícius Fontes wrote: When people release software under the GPL license, like Steve Underwood did with libunicall, spandsp and so on, they were supposed to know that other people has the right to use their code. The problem is that almost any licence term which tries to limit the obnoxious behaviour of other people has too many unpleasant side effects. GPL 2.0 is the best compromise I've found, so that is what I used for everything unless recently. To make my stuff licence compatible with FreeSwitch I recently relicenced most of my work as LGPL 2.1. This is having undesirable consequences, though. Its really a tough issue, and GPL 2.0 showed immense foresight in just accepting the non-existence of perfect solutions. GPL 3 seems to have forgotten the lesson somewhat. Most of the time I just want to give up producing anything at all. Steve So are you angry that he may gain monetarily from your your work, or is it hurt pride that he is basically taking credit for it? The answer to that should guide you in how you release your work in the future. Thanks, Steve Totaro I also want to add that if someone asked me to name the top five names that came to mind when thinking of Asterisk, Jim Dixon, Mark Spencer, Steve Underwood, Nicolas Gudino, and I will leave off the fifth as to not leave anybody out ;) Me, me, me! :D Or, Kevin, Russell, Olle, Josh, Critch (although he's been pretty quiet lately), I guess the list goes on. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVtVDQNt8rg0Kp4RAhlVAJ9L+JIUC3KC24Eptj00HCYW+/AMuwCfSfz/ RY5QyBXqBT12dWEW69EwKno= =fvT2 -END PGP SIGNATURE- ___ -- 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] asterisk + web services
On Jul 15, 2008, at 10:20 PM, Steve Edwards wrote: curl() doesn't fire up another process. The response is returned as just one big chunk. In my case, it was the HTML to an entire web page :) If you need to do a bunch of parsing, maybe an AGI calling libcurl -- saving a bunch of ugly dialplan. I guess I should have clarified... I make the output of the webservice a simple text string, with a delimiter not used in the results. If it's a whole HTML or XML response, I couldn't imagine using CURL. smime.p7s Description: S/MIME cryptographic signature ___ -- 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] OT: DNS security
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Alexander Lopez wrote: Snip On Wed, Jul 9, 2008 at 10:50 AM, C F [EMAIL PROTECTED] wrote: Very interesting article. I guess we won't know much more for another few weeks: http://www.breitbart.com/article.php?id=080709124916.zxdxcmkxshow_artic le=1 I thought this was common knowledge. I remember hearing about the flaw around 2000 or so. Thanks, Steve T Knowledge yes, but common, I don't think so. Cache Poisoning has been around since before 2000. I thought l0pht or someone did an article about it back around 2000. Then they went and spoke at Whitehouse dinners and stuff and kinda disappeared. In those days I was heavily into greyhat and IDS systems, but I'm pretty sure it was common knowledge. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfV6BDQNt8rg0Kp4RAt0cAJkBZCYDFO0vslMXpxnzjC2bVChxHgCgqiBg S/L936ORy9ubnvvYVjvaHVE= =klqT -END PGP SIGNATURE- ___ -- 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] Asterisk CAS connection to VConsole ISDN simulator
I'm attempting to get Asterisk to talk with a VConsole ISDN simulator that supports the following CAS protocols: CAS EM Wink Start FGD CAS EM Wink Start FGB I've tried configuring the Asterisk end with em_w, featb, featd, featdmf but with each of these, it either doesn't work at all, or I see calls coming in to Asterisk that shouldn't be, and unexpected robbed bit patterns at the simulator end. Has anyone else had experience connecting to a VConsole device, or failing that, can anyone point me towards the spec that each of these Asterisk protocols implements? Thanks, Mark ___ -- 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] Incoming
sip Thanks, --ag On Tue, Jul 15, 2008 at 8:39 PM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Artie Gold wrote: Folks: This is my first post, so please let me know if I transgress in any way... In updating to 1.4.21 recently, we've encountered a problem, when running over a satellite connection (where the latency is considerable; a regular internet connection did not exhibit this problem), where incoming calls are being dropped as a result of the sip handshake timing out (dropping down to 1.4.18.1 solved the problem for us). From reading the change logs and other posts, it seems that some work has been done in this area recently to get it right; it appears that, at least in the satellite case, things may have gotten a little too tight... If this rings a bell for anyone, any insight would be appreciated. These calls sip or iax? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIfVFWDQNt8rg0Kp4RAvj3AJ0bKXhQDS5v8bqAOQF9llPZdTh/wQCfclx3 vXJlSU/zoQY4mUxQhKE3mTY= =oFsV -END PGP SIGNATURE- ___ -- 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 -- Artie Gold F4W, Inc. ___ -- 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] Two way bandwidth test
Does anyone know of a bandwidth test that tests the upload with the download? All of the ones I can find will test the upload then the download. I from experience I have found that a 3M/768K DSL can only do about 256K/256K simultaneously. The only way I have of testing it is with FTP uploads and downloads or P2P sharing. I would like something more formal that would keep the upload speed the same as the download. VoIP as you know is symmetric. The one VoIP test I find doesn't tell you how many calls you can handle, just if it is VoIP ready. -Matt ___ -- 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] how to incorporate open hours
Hi All, I have got my voice menus setup. open hours and after hours. What do I have to code in the main menu to do the following. If between the hours of 9am - 5pm go to open hours All other hours go to after hours I've read all of the docs but don't quite understand it? Cheers 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
Re: [asterisk-users] how to incorporate open hours
What do I have to code in the main menu to do the following. If between the hours of 9am - 5pm go to open hours All other hours go to after hours You can do something like: exten = main switch no,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn) ___ -- 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] zap not getting callerid any more
Brian J. Murrell wrote: Unless you want to invest in a better card, you may just have to live with the problem. Which means what, a multiport and multi-hundreds of dollar card? I'm just a home user. I don't have hundreds of dollars to spend on a single piece of phone hardware. I hadn't realised this was for a home server... yes I agree, for a home server the Digium or Sangoma cards are a little too expensive. I wonder how much using something like an SPA-3102 (with both an FXS and FXO ports) eliminates these problems and brings reliability to the table. I can't speak for the SPA-3102, but the SPA-3000 I use here at home doesn't do a brilliant job. I suffer from varying amounts of echo on many calls, though oddly enough I've pretty much learnt to ignore it. :) As for Caller ID support, I can't help there - it /does/ support Caller ID, but since I don't really want to pay an extra $6 or so just to get caller ID on a number that usually rings only when telemarketers get around to it, I have no idea how reliable it is. I'm sure you'll find it amusing to find out that I was toying with the idea of getting an X100 card for my server. :) Unfortunately, they appear to be very hard to find in Australia. ___ -- 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] how to incorporate open hours
OK Ive done this. exten=7000,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn) 7000 is the extension of main menu Where do I put the reference to open hours menu in the statement above. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: Wednesday, 16 July 2008 3:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] how to incorporate open hours What do I have to code in the main menu to do the following. If between the hours of 9am - 5pm go to open hours All other hours go to after hours You can do something like: exten = main switch no,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn) ___ -- 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
