[asterisk-users] Use one ring-group for ISN truncs
Hi, A question. We are using TE420 cards. Normally we configure for each truncs one ring-group. group=1 channel = 1-15,17-31 group=2 channel = 32-46,48-62 group=3 channel = 63-77,79-93 group=4 channel = 94-108,110-124 My question now, is it possible to join more ring-groups to one ring-group? Example: Group 1 channel = 1-15,17-31 channel = 32-46,48-62 group=2 channel = 63-77,79-93 channel = 94-108,110-124 Regards, Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use one group for ISN truncs
Hi, A question. We are using TE420 cards. Normally we configure for each truncs one group. group=1 channel = 1-15,17-31 group=2 channel = 32-46,48-62 group=3 channel = 63-77,79-93 group=4 channel = 94-108,110-124 My question now, is it possible to join more groups to one group? Example: Group 1 channel = 1-15,17-31 channel = 32-46,48-62 group=2 channel = 63-77,79-93 channel = 94-108,110-124 We are using the group number for the dial en originate command. For example: Zap/g3/0612345678 Regards, Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use one group for ISN truncs
On Mon, Jun 28, 2010 at 09:16:37AM +0200, Arjan Kroon | Mobillion wrote: Hi, A question. We are using TE420 cards. Normally we configure for each truncs one group. group=1 channel = 1-15,17-31 group=2 channel = 32-46,48-62 group=3 channel = 63-77,79-93 group=4 channel = 94-108,110-124 My question now, is it possible to join more groups to one group? Example: Group 1 channel = 1-15,17-31 channel = 32-46,48-62 group=2 channel = 63-77,79-93 channel = 94-108,110-124 Err... sure: group=1 channel = 1-15,17-31 group=1 channel = 32-46,48-62 group=2 channel = 63-77,79-93 group=2 channel = 94-108,110-124 Or: group=1 channel = 1-15,17-31 channel = 32-46,48-62 group=2 channel = 63-77,79-93 channel = 94-108,110-124 Or: group=1 channel = 1-15,17-31,32-46,48-62 group=2 channel = 63-77,79-93,94-108,110-124 We are using the group number for the dial en originate command. For example: Zap/g3/0612345678 For obvious reasons, this will not work on Asterisk versions = 1.6.0 [1] Note you can use something along the lines of: group=1,11 channel = 1-15,17-31 group=2,11 channel = 32-46,48-62 group=3,12 channel = 63-77,79-93 group=4,12 channel = 94-108,110-124 In this case you can dial throgh g1, which will use channels of the first port, or g11, which will use channels of the first two ports. [1] The ChannelType Zap. Use DAHDI. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem attended transfer with ilbc
I have an Asterisk server on our LAN that serves our office VOIP phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are ulaw/alaw We use attended transfer extensively. If our trunk is ulaw/alaw they work fine. If the trunk is ilbc we have problems 1- incoming PSTN call routed via voipfone SIP down the trunk to our server 2- our phones ring ok, caller can be answered (e.g. by A) 3- A requests attended transfer to another phone (B) on the LAN- incoming caller put on hold, A can talk to B, B can talk to A 4- A hangs up, B is connected to caller. B can hear caller, but caller cannot hear B. Console output: Asked to transmit frame type 64, while native formats is 0x400 (ilbc)(1024) read/write = 0x40 (slin)(64)/0x400 (ilbc)(1024) Running Asterisk 1.6.2.9 on Ubuntu Karmic- self compiled (do not seem to be able to compile deb source package with ilbc, and deb package does not have ilbc) Any idea what may be happening? John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem attended transfer with ilbc
On Mon, Jun 28, 2010 at 5:15 AM, John Taylor j...@vetsurgeon.org.uk wrote: Any idea what may be happening? acknowledged https://issues.asterisk.org/view.php?id=16287 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip add header
It seems that for local channels (asterisk 1.4.33) the variable Variable: SIPADDHEADER=Alert-Info: Ring Answer (call polycom phones and ring then auto answer) Is ignored, Is this just an oversite or is there some reason? It works fine with I call the SIP phone directly - however - when I first call the Local channel - then Dial the SIP phone the SIPADDHEADER doesnt seem to do anything. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip add header
On 28 Jun 2010, at 13:08, Jerry Geis wrote: It works fine with I call the SIP phone directly - however - when I first call the Local channel - then Dial the SIP phone the SIPADDHEADER doesnt seem to do anything. Are you adding the header before or after you dial the local channel? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call drops on group paging asterisk - 1.4.22.1
Thanks Mike! We are using one Aastra phone with expansion module and the remaining 27 phones are from Yealink (new phones that came out), currently Aastra phone used to freeze while paging, but now we replaced the aastra to Yealink and will see if this solves the problem. Sandesh On Fri, Jun 25, 2010 at 12:02 PM, Mike l...@net-wall.com wrote: The phone brand and model might matter here, I have had no such problems with Polycom phones. Mike *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *das sandesh *Sent:* Friday, June 25, 2010 12:58 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call drops on group paging asterisk - 1.4.22.1 Hi All, We are using group paging and our asterisk version: 1.4.22.1, but when ever any one page to the whole group (28 extensions), the calls which are on hold on those extensions will be dropped, is there any other way to have this feature or to go with Overhead paging. Currently this has become a serious problem, can anyone through some light on this group paging senario? Thank you very much Regards Sandesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem attended transfer with ilbc
Paul Belanger schrieb: On Mon, Jun 28, 2010 at 5:15 AM, John Taylor j...@vetsurgeon.org.uk wrote: Any idea what may be happening? acknowledged https://issues.asterisk.org/view.php?id=16287 hello, i´ve reported the same bug i´ve found out later with this issue: https://issues.asterisk.org/view.php?id=17400 The problem is the answer before the dial. If you didnt use this you wont have this problem. iam still searching a solution to get the right indication value so this doesnt happens. best regards. steve -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup a ringing Queue member
Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192type=bug' -O - | patch -p0/* ?? Does this mean I have a patched asterisk ? (I ask this because some applications require a non-patched asterisk version) Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup a ringing Queue member
On Mon, Jun 28, 2010 at 10:00 AM, Jonas Kellens jonas.kell...@telenet.be wrote: I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : wget 'https://issues.asterisk.org/file_download.php?file_id=17192type=bug' -O - | patch -p0 ?? This patch was merged in Asterisk 1.4.17, so you are already running it. Does this mean I have a patched asterisk ? (I ask this because some applications require a non-patched asterisk version) Yes. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup a ringing Queue member
Does this mean I have a patched asterisk ? (I ask this because some applications require a non-patched asterisk version) Yes. What is then the unpatched version of Asterisk 1.4.30 ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pickup a ringing Queue member
On 28 Jun 2010, at 15:36, Jonas Kellens wrote: Does this mean I have a patched asterisk ? (I ask this because some applications require a non-patched asterisk version) Yes. What is then the unpatched version of Asterisk 1.4.30 ?? The one you have before you apply the patch?.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Never seen Problem !!!
One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0. Today, when they downloaded , the CDR from the carrier site for 26th June 2010 , they see 50% calls are NEVER dialed by Dialer but it appears in CDR. Amazingly, all the call durations are of 29-30 secs. When we checked the status of the same in Dialer, lead is present there but its marked as NEW which means Dialer has ever dialed those calls. How can that happen ? My carrier says that my dialer sent INVITE to the server. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Handling DTMF for number 4
Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK except for the DTMF for number 4, because if the caller from mobile dial 1004 or 1014 extensions -which have the number 4- the calls are errouneosly established with extension 1000. I live in Argentina, but I don't know if the DTMF frequencies are the same than other countries or I have to make a change in somewhere. Can be a problem with the detection of DTMF for number 4 in Asterisk ??? Thanks a lot Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling DTMF for number 4
Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK except for the DTMF for number 4, because if the caller from mobile dial 1004 or 1014 extensions -which have the number 4- the calls are errouneosly established with extension 1000. I live in Argentina, but I don't know if the DTMF frequencies are the same than other countries or I have to make a change in somewhere. Can be a problem with the detection of DTMF for number 4 in Asterisk ??? Thanks a lot Alejandro The gsm gateway would be performing the DTMF detection and just sending on what it detected as you have the DTMFCODE set as RFC. Maybe if you set the DTMF mode to INBAND it may pass the audio straight through and allow asterisk to detect it. The problem is that mobile calls are heavily compressed so any entered digits are converted to tones once the information reaches the network operator. As the call is gong back over a mobile connection the DTMF is compressed which results in unreliable detection. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling DTMF for number 4
Thanks Gareth, when you say that I can choose INBAND for DTMF MODE in the GSM Gateway, that implies that the DTMF MODE of the Asterisk extension registered for the GSM Gateway has to be set to INBAND too or can it remain in RFC2238 ??? Because I have all my Asterisk extensions and IP telephones set up with DTMFMODE = RFC2238 by now, and I can't understand if you suggest me I change the DTMFMODE from RFC2238 to INBAND just in the GSM Gateway or everywhere. Thanks again. 2010/6/28 Gareth Blades list-aster...@skycomuk.com: Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK except for the DTMF for number 4, because if the caller from mobile dial 1004 or 1014 extensions -which have the number 4- the calls are errouneosly established with extension 1000. I live in Argentina, but I don't know if the DTMF frequencies are the same than other countries or I have to make a change in somewhere. Can be a problem with the detection of DTMF for number 4 in Asterisk ??? Thanks a lot Alejandro The gsm gateway would be performing the DTMF detection and just sending on what it detected as you have the DTMFCODE set as RFC. Maybe if you set the DTMF mode to INBAND it may pass the audio straight through and allow asterisk to detect it. The problem is that mobile calls are heavily compressed so any entered digits are converted to tones once the information reaches the network operator. As the call is gong back over a mobile connection the DTMF is compressed which results in unreliable detection. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling DTMF for number 4
It would need to be set in the gsm gateway and in the corresponding section in sip.conf which connects to that gsm gateway. Everything else should be left as rfc. It may help or it might not. The gateway might also have some settings you can change to improve the detection. The Patton unit I have allows you to change the codec rx gain which can help dtmf detection. I only set it up today so havent really tested it yet. Alejandro Cabrera Obed wrote: Thanks Gareth, when you say that I can choose INBAND for DTMF MODE in the GSM Gateway, that implies that the DTMF MODE of the Asterisk extension registered for the GSM Gateway has to be set to INBAND too or can it remain in RFC2238 ??? Because I have all my Asterisk extensions and IP telephones set up with DTMFMODE = RFC2238 by now, and I can't understand if you suggest me I change the DTMFMODE from RFC2238 to INBAND just in the GSM Gateway or everywhere. Thanks again. 2010/6/28 Gareth Blades list-aster...@skycomuk.com: Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for mobile phone calls coming from a GSM Gateway. All the components are set up in DTMFMODE = RFC2238, and so when the caller from mobile touches the IP phone LAN extension, the call is succesfully established. Everything is OK except for the DTMF for number 4, because if the caller from mobile dial 1004 or 1014 extensions -which have the number 4- the calls are errouneosly established with extension 1000. I live in Argentina, but I don't know if the DTMF frequencies are the same than other countries or I have to make a change in somewhere. Can be a problem with the detection of DTMF for number 4 in Asterisk ??? Thanks a lot Alejandro The gsm gateway would be performing the DTMF detection and just sending on what it detected as you have the DTMFCODE set as RFC. Maybe if you set the DTMF mode to INBAND it may pass the audio straight through and allow asterisk to detect it. The problem is that mobile calls are heavily compressed so any entered digits are converted to tones once the information reaches the network operator. As the call is gong back over a mobile connection the DTMF is compressed which results in unreliable detection. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Never seen Problem !!!
Let me know if you need any further info !! On Mon, Jun 28, 2010 at 9:15 PM, G M gm.cu...@gmail.com wrote: One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0. Today, when they downloaded , the CDR from the carrier site for 26th June 2010 , they see 50% calls are NEVER dialed by Dialer but it appears in CDR. Amazingly, all the call durations are of 29-30 secs. When we checked the status of the same in Dialer, lead is present there but its marked as NEW which means Dialer has ever dialed those calls. How can that happen ? My carrier says that my dialer sent INVITE to the server. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip server
Hi, Can i use asterisk as sip server for manage call Transmission between gateways Best Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip server
Yes CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mohamed daif Sent: Monday, June 28, 2010 2:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip server Hi, Can i use asterisk as sip server for manage call Transmission between gateways Best Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 and multiple parking
Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different lots so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same question. Is there some sort of sample conf file of how I would get this functionnal on the latest 1.6.x? Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip server
hi i want to use asterisk as a sip server without installing any hardware in this machine the question is how can i configure the external getaways with asterisk how can i configure the costumer who is i provide calls to hem what is the billing software can i use to calculate the the calls and manage the rate -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with TE411P and DAHDI
We just recently upgraded a server from Zaptel to DAHDI and Asterisk 1.4.30 to 1.6.2.9 and now we are getting this message before the server reboots every few minutes: Message from syslogd@ at Mon Jun 28 15:17:48 2010 ... pbx kernel: Dazed and confused, but trying to continue Jun 28 15:17:48 pbx kernel: Uhhuh. NMI received for unknown reason 35 on CPU 0. Jun 28 15:17:48 pbx kernel: Do you have a strange power saving mode enabled? Jun 28 15:17:48 pbx kernel: Dazed and confused, but trying to continue This is an IBM server that has been in operation for over three years now. It is running CentOS 5.5 with all the updates applied. The upgrade was made last week on Tuesday and thee problem started on Saturday when it rebooted a couple times but today it is rebooting every few minutes. Any ideas on how to diagnose the problem? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip server
On Mon, Jun 28, 2010 at 4:02 PM, mohamed daif mohamed.d...@gmail.com wrote: i want to use asterisk as a sip server without installing any hardware in this machine the question is how can i configure the external getaways with asterisk how can i configure the costumer who is i provide calls to hem what is the billing software can i use to calculate the the calls and manage the rate Time to do some reading: http://astbook.asteriskdocs.org/ -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 and multiple parking
Hello there You should have a look at features.conf Regards Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike Sendt: 28. juni 2010 21:39 Til: 'Asterisk Users Mailing List - Non-Commercial Discussion' Emne: [asterisk-users] Asterisk 1.6 and multiple parking Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different lots so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same question. Is there some sort of sample conf file of how I would get this functionnal on the latest 1.6.x? Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TE411P and DAHDI
On 06/28/2010 03:27 PM, Carlos Chavez wrote: We just recently upgraded a server from Zaptel to DAHDI and Asterisk 1.4.30 to 1.6.2.9 and now we are getting this message before the server reboots every few minutes: Message from syslogd@ at Mon Jun 28 15:17:48 2010 ... pbx kernel: Dazed and confused, but trying to continue Jun 28 15:17:48 pbx kernel: Uhhuh. NMI received for unknown reason 35 on CPU 0. Jun 28 15:17:48 pbx kernel: Do you have a strange power saving mode enabled? Jun 28 15:17:48 pbx kernel: Dazed and confused, but trying to continue This is an IBM server that has been in operation for over three years now. It is running CentOS 5.5 with all the updates applied. The upgrade was made last week on Tuesday and thee problem started on Saturday when it rebooted a couple times but today it is rebooting every few minutes. Any ideas on how to diagnose the problem? If your card is still under warranty, you can may want to contact customer support for assistance. However, the first question I have (in order to rule out any hardware failures that just happen to coincide with your update) is if you revert the the previous version of Zaptel and Asterisk and leave the kernel / platform version to same, is the system stable again? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip add header
At 8:08 AM on 28 Jun 2010, Jerry Geis wrote: It seems that for local channels (asterisk 1.4.33) the variable Variable: SIPADDHEADER=Alert-Info: Ring Answer (call polycom phones and ring then auto answer) Is ignored, Is this just an oversite or is there some reason? It works fine with I call the SIP phone directly - however - when I first call the Local channel - then Dial the SIP phone the SIPADDHEADER doesnt seem to do anything. Have you tried adding an underscore? Set(_SIPADDHEADER=Alert-Info: Ring Answer) Without the underscore, the variable won't be inherited by the Local channel. Also, look up the /n option to the Local channel. That may affect it, but I can't say how off the top of my head. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call file structure and syntax
Well, I¹ve tried this, and something just isn¹t right. Here¹s the context
from dialplan show, so I know it¹s loaded anyway:
[ Context 'PressTwo' created by 'pbx_config' ]
'*' =1. Goto(accept|s|1) [pbx_config]
'1' =1. ForkCDR(v,s(fullcmd=${Data}))[pbx_config]
2. Background(${Data}) [pbx_config]
3. Background(repeatmsg)[pbx_config]
4. WaitExten(5,m) [pbx_config]
5. Hangup() [pbx_config]
'2' =1. Background(calllater)[pbx_config]
2. ForkCDR(v,s(reject=${Data})) [pbx_config]
3. Hangup() [pbx_config]
'3' =1. Goto(accept|1|2) [pbx_config]
'i' =1. Goto(accept|s|1) [pbx_config]
's' =1. Answer() [pbx_config]
2. Background(important)[pbx_config]
3. WaitExten(5,m) [pbx_config]
't' =1. Goto(accept|s|1) [pbx_config]
I (hopefully correctly) translated your dialplan into a simple AMI command
set thus:
Action: Originate
Channel: SIP/ShoreTel-1
Variable: Data=/var/lib/asterisk/sounds/custom/msg1.wav
Context: PressTwo
priority: 1
Number: 7979
When I hit AMI via telnet, login, and execute the above, here's the output:
Response: Error
Message: Originate failed
Event: Newchannel
Privilege: call,all
Channel: SIP/ShoreTel-1-0004
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: from-internal
Uniqueid: 1277768352.7
Event: VarSet
Privilege: dialplan,all
Channel: SIP/ShoreTel-1-0004
Variable: SIPCALLID
Value: 39251bb451a38e136293d70252d0d...@10.10.6.45
Uniqueid: 1277768352.7
Event: VarSet
Privilege: dialplan,all
Channel: SIP/ShoreTel-1-0004
Variable: Data
Value: /var/lib/asterisk/sounds/custom/msg1.wav
Uniqueid: 1277768352.7
Event: NewAccountCode
Privilege: call,all
Channel: SIP/ShoreTel-1-0004
Uniqueid: 1277768352.7
AccountCode:
OldAccountCode:
Event: NewCallerid
Privilege: call,all
Channel: SIP/ShoreTel-1-0004
CallerIDNum:
CallerIDName:
Uniqueid: 1277768352.7
CID-CallingPres: 0 (Presentation Allowed, Not Screened)
Event: Hangup
Privilege: call,all
Channel: SIP/ShoreTel-1-0004
Uniqueid: 1277768352.7
CallerIDNum: unknown
CallerIDName: unknown
Cause: 17
Cause-txt: User busy
Event: RTPReceiverStat
Privilege: reporting,all
SSRC: 0
ReceivedPackets: 0
LostPackets: 0
Jitter: 0.
Transit: 0.
RRCount: 0
Event: RTPSenderStat
Privilege: reporting,all
SSRC: 462309403
SentPackets: 0
LostPackets: 0
Jitter: 0
SRCount: 0
RTT: 0.00
Event: RTPReceiverStat
Privilege: reporting,all
SSRC: 0
ReceivedPackets: 0
LostPackets: 0
Jitter: 0.
Transit: 0.
RRCount: 0
Event: RTPSenderStat
Privilege: reporting,all
SSRC: 1486119401
SentPackets: 0
LostPackets: 0
Jitter: 0
SRCount: 0
RTT: 0.00
Thing is, I know I can dial out via that SIP trunk, and it's a test system
nobody else is using, so why am I getting User busy here?
On 6/22/10 10:31 AM, Danny Nicholas da...@debsinc.com wrote:
#1 once you¹ve got to this point, AMI would be a better option than a call
file
#2 - using AMI or a call file, you are going to want to use the context-based
method instead of application to get the most ³bang for your buck²
I use a bigger instance of this to play a message and accept 1 or 2 from the
user
; this context is used by AMI to play a message
[accept]
exten = s,1,Answer
exten = s,n,Background(important)
exten = s,n,WaitExten(5,m)
exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))
exten = 1,n,Background(${Data})
exten = 1,n,Background(repeatmsg)
exten = 1,n,WaitExten(5,m)
exten = 1,n,Hangup
exten = 2,1,Background(calllater)
exten = 2,n,ForkCDR(v,s(reject=${Data}))
exten = 2,n,Hangup
exten = 3,1,Goto(accept|1|2)
exten = *,1,Goto(accept|s|1)
exten = i,1,Goto(accept|s|1)
exten = t,1,Goto(accept|s|1)
here¹s the call file
Action = 'Originate',
Channel = DAHDI/1,
Variable = Data=/tmp/test.gsm²,
Exten = 'SIP/170',
Context = 'accept',
priority = 1,
Number = 5551212
Using the accept context, 5551212 is called on DAHDI/1 and user hears
important.gsm. then they press 1 to hear test.gsm or 2 to hear it later.
Hope this is helpful
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
[asterisk-users] restricting sip users to a certain useragent
Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM Customer Relations Management with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will prevent the agents we are restricting them to only be able to dial through the software which is already restricted to their seats in the call center.. but someone might sniff around .. and get the sip username and password assigned to him and use it through Zoiper or any other softphone to make calls ..our agents are allowed international calls .. so we want to restrict them to only use our dialer.Is that possible?Asterisk version 1.4.33regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call file structure and syntax
Well, I¹ve tried this, and something just isn¹t right. Look here: Event: Hangup Channel: SIP/ShoreTel-1-0004 Cause: 17 Cause-txt: User busy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Update the LCD with the callee's name after dialing
Is is possible with a Polycom phone to update the LCD with the callee's name after dialing them? When you dial ext 103 now, it says 'To:103'...would be nice if could have 'To:Dan Marino' This is the case even when you have a contact for ext 103. None of the phones I have ever tested do this, Polycom, Linksys, Cisco, Grandstream, Yealink, etc. -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restricting sip users to a certain useragent
This is a very good question. I faced the same problem some time ago, and by
goggling found out that somebody had actually programmed a patch for this
purpose, but it never got approved to go into the main branch of Asterisk.
If you google, you'll probably found out details on it.
I am, however, found a very simple way to do it, which is not perfectly
secure because it is crackable by any hi tech cracker who could send
modified SIP headers, but otherwise it does its job great.
Here is how you can do it (I use Asterisk 1.4):
First check the user agent name using sip show peers. Lets say sip user
agent is Sipura/SPA1001-3.1.8(SEc).
When a SIP phone registers on your asterisk server, asterisk stores its user
agent information in variable ${SIPUSERAGENT}. This means when an extension
dials out, you can check in the dialplan if the ${SIPUSERAGENT} matches your
user agent or not, and based on the result you can decide how to proceed
with the call.
I use AEL, and it'll look like this:
// Check if the user agent is the one we supplied
if (${SIPUSERAGENT}!=Sipura/SPA1001-3.1.8(SEc)) {
// If not, hangup the call or do something else. I block the IP in
iptables and record IP in MySQL for future reference
NoCDR();
Hangup();
}
// Otherwise continue the dialplan
In regular non AEL config, which is harder to work with, it will look
something like this:
exten = _NXXNXX,1,...
exten = _NXXNXX,n,...
...
...
...
exten =
_NXXNXX,n,GotoIf($[${SIPUSERAGENT}!=Sipura/SPA1001-3.1.8(SEc)]?hangup:continue)
exten = _NXXNXX,n(hangup),NoCDR()
exten = _NXXNXX,n,Hangup()
exten = _NXXNXX,n(continue),...
exten = _NXXNXX,n,...
As I said earlier, if an experienced cracker really wants, and knows what
sip user agent name you are using, he can figure out that you are checking
the sip user agent name, and then send custom sip user agent name with the
same name. But in your case its probability is very little. In fact I was
using a webphone with free calls to anywhere in North American and some
other countries right from my website, and wouldn't really care if crackers
would crack it or not, because call duration was only one minute, but this
setup really helped block a lot of crakcers, because before they would know
I was checking extensions by user agent name, their IP address would already
be blocked.
--
Zeeshan
On Mon, Jun 28, 2010 at 7:58 PM, Tarek Sawah tareksa...@hotmail.com wrote:
Greetings list,
this question is rather a pain in my side.. i have been trying to figure it
out.. it could be simple.
i have a customer with a callcenter .. we developed a CRM
Customer Relations Management with an SIP dialers built in.
the question is the following.. is it possible to force the agents (users)
to use a certain UserAgent which is the one built-in our system? this way
will prevent the agents we are restricting them to only be able to dial
through the software which is already restricted to their seats in the call
center.. but someone might sniff around .. and get the sip username and
password assigned to him and use it through Zoiper or any other softphone to
make calls ..our agents are allowed international calls .. so we want to
restrict them to only use our dialer.
Is that possible?
Asterisk version 1.4.33
regards
-- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1
386 492 9993
--
The New Busy is not the too busy. Combine all your e-mail accounts with
Hotmail. Get
busy.http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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--
Zeeshan A Zakaria
--
_
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Re: [asterisk-users] Update the LCD with the callee's name after dialing
Remote Party ID in trunk, it works There are hacks for other versions. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Mon, Jun 28, 2010 at 8:40 PM, Matt Darnell mattdarn...@gmail.com wrote: Is is possible with a Polycom phone to update the LCD with the callee's name after dialing them? When you dial ext 103 now, it says 'To:103'...would be nice if could have 'To:Dan Marino' This is the case even when you have a contact for ext 103. None of the phones I have ever tested do this, Polycom, Linksys, Cisco, Grandstream, Yealink, etc. -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What‘s the best operating sys tem suggest for Asterisk 1.6.2.9
hi, list i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. i want to use CentOS5.2 or CentOS 5.4. Which is better and stable? Thanks for your help. -- Thanks for your supporting, have a nice day. Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What are the guts of AsteriskNOW and how it compares to other popular flavors available?
Hi Everyone, I want to know a bit about the guts of the current AsterisNOW system. I know that FreePBX is embraced as the main GUI but is just an install of CentOS 5.4 + (Asterisk/FreePBX from Yum repos)? - Or is there anymore to this? Maybe some security tools? - Or is Asterisk built from the source? - Is there some other program installed to facilitate a better PBX system than those of the competitor flavours (e.g. Trixbox, piaf, Elastix)? or does it generally fall behind those? - I would be very happy if you can point me to section of the image file which is responsible for post install of CentOS as I think that is the approach to get an AsteriskNOW system running with Asterisk and etc... Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to configure key sequence in features.conf.
Hi all,
I need to achieve the following function:
user 1 call to user 2, In the process they calling, if user 2 press *3 keys,
then the call hangup and playback voice file.
My setting as following:
* features.conf**
[featuremap]
textkey1 = *3
[applicationmap]
testkey1 = *3,callee,Playback,demo-instruct
featuredigittimeout = 2
atxfernoanswertimeout = 15
*sip.conf
[1]
type=friend
username=1
secret=1
host=dynamic
context=from-sip
[2]
type=friend
username=2
secret=2
host=dynamic
context=from-sip
extensions.conf**
[from-sip]
include = testkey1
include = keycommand
exten = 1,1,Dial(sip/1)
exten = 1,2,Hangup()
exten = 2,1,Dial(sip/2)
exten = 2,2,Hangup()
[keycommand]
exten = s,1,SoftHangup(${BRIDGEPEER})
***
when I use sip extensions 1 call to sip extensions 2, In the process they
calling, i press *3 keys. But there is nothing happened, the call still
through, and the system does not play the voice file.
I don't know where is the problem lies in. Can anybody help me ?
Thank you!
Louis.
--
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