Re: [asterisk-users] PRI timing settings
On 2014-08-19 23:56, Jeff LaCoursiere wrote: Hello, I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. The telco has been working with their switch manufacturer and took the output of pri show span 1 from me and came back with this: quote--- Please check your timers below. How did you determine your settings? * Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T201: 1000 T202: 2000 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000Our switch: Telcordia National ISDN 2: Range 10 - 90 seconds, default 90 seconds. * T312: 6000 T313: 4000 T316: -1Our switch: Telcordia National ISDN 2: Range 10 - 120 seconds, default 30 seconds. N316: 2 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 ---end quote--- Now I have no idea what T309 or T316 represent, but it seems odd that timers and counters would cause such an odd result... and the failure is immediate, not after some amount of timeout. Can anyone shed light on these settings and tell me if they are configurable? I don't think that this has something to do with the timers. I recommend to enable debug of your pri connection to get a trace to a sprint phone number. There you should get the disconnect cause of the call. Then your provider should be able to help you out. Maybe the reason of the problem is an incorrect number setting. Regards Hans -- _ -- 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] log caller hangup events
Logically yes, once the call hangup, the hangup handler will execute. Regards, On Mon, Aug 18, 2014 at 7:04 PM, Paul Greenberg p...@greenberg.pro wrote: Hi, I am mostly concerned with inbound calls. Would it work the same? Regards, Paul -- *From:* asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnan N gopalakrishnan...@gmail.com *Sent:* Monday, August 18, 2014 4:13 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] log caller hangup events Hi, You can use Hangup handler. May be this post can you help you, http://gblades.blogspot.in/2013/07/how-to-get-sip-response-code-in.html Regards On Mon, Aug 18, 2014 at 9:45 AM, Paul Greenberg p...@greenberg.pro wrote: All, I would like to log a message whenever a party hangs up a call or session, i.e. no Dial(), user drops off a menu. The message should include the length of the user's session, the session's start time, and called ID. Theoretically, I could set up a channel variable and then ... Any advice would be most welcome! Regards, Paul -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Asterisk and BT Versatility PBX via NT BRI port
On 08/04/2014 03:03 PM, David Duffett wrote: Please come back to let us know if this actually does fix the issue. So far so good the external voltage supply for the OpenVOX card has arrived and I can confirm that the BT Versatility PBX worked like a charm. DAHDI channel has been configured in PtP mode. Cheers, Roberto Fichera. On 4 Aug 2014 14:36, Roberto Fichera ker...@tekno-soft.it mailto:ker...@tekno-soft.it wrote: On 08/04/2014 01:21 PM, David Duffett wrote: If the OpenVox card can supply the voltage, then it will a configuration option (probably in hardware, like some jumpers) of the card itself. Yep! The OpenVOX card has a jumper for each port and a connector where to insert the external voltage supply device. I was going to point you to the Xorcom Astribank, which I know supplies the required voltage. Ah! Ok! I was thinking you was giving me something like a temporary solution for supply the voltage to the PBX. Cheers, Roberto Fichera. All the best, David On 4 Aug 2014 13:16, Roberto Fichera ker...@tekno-soft.it mailto:ker...@tekno-soft.it wrote: On 08/03/2014 04:37 PM, David Duffett wrote: Does the BT Versatility system you are trying to connect require a voltage on the line, which it would normally get from the BT connection? Some BRI equipment does, and some does not, and it may be the Panasonic system you refer to is happy without the voltage, while, perhaps, the BT Versatility needs it. If you find out that it does need the voltage, I do not think the OpenVox card you mentioned will supply it, but I can point you to a solution if required. Just checked if the BT Versatility takes or not the voltage from the NT-1 and it does! So the problem looks really this! Can you point me to how supply voltage to the PBX while I'm waiting the given OpenVox adapter being delivered? Cheers, Roberto Fichera. All the best, On Sunday, August 3, 2014, Mc GRATH Ricardo mcgra...@mail2web.com mailto:mcgra...@mail2web.com wrote: Hi, it seems have problem with physical issue layer 1, first check wiring connection, by the way you can check with card led (If ISDN plugs into the port, the LED will not blink, but in red color.) after dahdi alarm status, and dahdi restart command. At last should be signalling mode to signalling = bri_cpe_ptmp Good luck Mc GRATH Ricardo E-Mail mcgra...@mail2web.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 -- Digium logo *David Duffett* Digium, Inc. · Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK direct/fax: +1 256 428 6119 · mobile: +44 7722 442236 twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett http://www.linkedin.com/in/davidduffett Check us out at: http://digium.com http://digium.com/ · http://asterisk.org http://www.asterisk.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Connecting Asterisk and BT Versatility PBX via NT BRI port
Great news! On Wednesday, August 20, 2014, Roberto Fichera ker...@tekno-soft.it wrote: On 08/04/2014 03:03 PM, David Duffett wrote: Please come back to let us know if this actually does fix the issue. So far so good the external voltage supply for the OpenVOX card has arrived and I can confirm that the BT Versatility PBX worked like a charm. DAHDI channel has been configured in PtP mode. Cheers, Roberto Fichera. On 4 Aug 2014 14:36, Roberto Fichera ker...@tekno-soft.it javascript:_e(%7B%7D,'cvml','ker...@tekno-soft.it'); wrote: On 08/04/2014 01:21 PM, David Duffett wrote: If the OpenVox card can supply the voltage, then it will a configuration option (probably in hardware, like some jumpers) of the card itself. Yep! The OpenVOX card has a jumper for each port and a connector where to insert the external voltage supply device. I was going to point you to the Xorcom Astribank, which I know supplies the required voltage. Ah! Ok! I was thinking you was giving me something like a temporary solution for supply the voltage to the PBX. Cheers, Roberto Fichera. All the best, David On 4 Aug 2014 13:16, Roberto Fichera ker...@tekno-soft.it javascript:_e(%7B%7D,'cvml','ker...@tekno-soft.it'); wrote: On 08/03/2014 04:37 PM, David Duffett wrote: Does the BT Versatility system you are trying to connect require a voltage on the line, which it would normally get from the BT connection? Some BRI equipment does, and some does not, and it may be the Panasonic system you refer to is happy without the voltage, while, perhaps, the BT Versatility needs it. If you find out that it does need the voltage, I do not think the OpenVox card you mentioned will supply it, but I can point you to a solution if required. Just checked if the BT Versatility takes or not the voltage from the NT-1 and it does! So the problem looks really this! Can you point me to how supply voltage to the PBX while I'm waiting the given OpenVox adapter being delivered? Cheers, Roberto Fichera. All the best, On Sunday, August 3, 2014, Mc GRATH Ricardo mcgra...@mail2web.com javascript:_e(%7B%7D,'cvml','mcgra...@mail2web.com'); wrote: Hi, it seems have problem with physical issue layer 1, first check wiring connection, by the way you can check with card led (If ISDN plugs into the port, the LED will not blink, but in red color.) after dahdi alarm status, and dahdi restart command. At last should be signalling mode to signalling = bri_cpe_ptmp Good luck Mc GRATH Ricardo E-Mail mcgra...@mail2web.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 -- [image: Digium logo] *David Duffett* Digium, Inc. · Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK direct/fax: +1 256 428 6119 · mobile: +44 7722 442236 twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com · http://asterisk.org http://www.asterisk.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] *David Duffett* Digium, Inc. · Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK direct/fax: +1 256 428 6119 · mobile: +44 7722 442236 twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com · http://asterisk.org http://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] PRI timing settings
On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.netmailto:j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don’t know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don’t know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl -- _ -- 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] PRI timing settings
On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net mailto:j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don’t know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don’t know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] PRI timing settings
For my NI2 PRIs I've always used 10 digits for everything and no +1 --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, August 20, 2014 9:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI timing settings On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don't know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don't know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] PRI timing settings
NXXNXX is the correct format of CallerID numbers in NANPA. The leading 1 is not part of any NANPA phone number. Toll free area codes are also not valid for CallerID. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, August 20, 2014 2:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI timing settings On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.netmailto:j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don't know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don't know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] customizing internal calls
We have a client set up with an Elastix 2.5 install running Asterisk 11 and 2.11 using Aastra 6731i IP phones. The client would like to have the feature where all internal calls use a distinctive ring (alert-info) so it sounds different than incoming external calls and they want it to check if the extension being called is on another call. If they are on another call, the client would like an audio message to be played telling them that extension is on another call, but continue to ring (as call waiting is enabled on all extensions). Example: Extension 1002 is on a call with a vendor. Extension 1001 dials extension 1002 and gets Bellcore-dr3 instead of Bellcore-dr1 followed by an audio file stating “this person is currently on another call”. The phone continues to ring and goes to their busy message at which time extension 1001 leaves a voice message for extension 1002. David Shauger Vice President 678-317-9444 x5 - voice 404-886-7603 - cell This email has been certified by Comodo Email certification helps prevent identity theft -- _ -- 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] PRI timing settings
What about the text portion? Should that never be sent? I was indeed sending the '1', and I will remove that to see if it solves my problem, but I also have the company name in there. I feel like a newb asking such questions, but I've never had this issue before :) Company 1NXXNXX Cheers, j On 08/20/2014 09:46 AM, Eric Wieling wrote: NXXNXX is the correct format of CallerID numbers in NANPA. The leading 1 is not part of any NANPA phone number. Toll free “area codes” are also not valid for CallerID. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere *Sent:* Wednesday, August 20, 2014 2:41 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI timing settings On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net mailto:j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don’t know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don’t know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] PRI timing settings
CallerID Name doesn't really matter. Either your carrier will remove it when handing the call off to the next hop or the terminating carrier will ignore any CallerID name data and do a name lookup in their own database using the CallerID number. This is why your CallerID name can be different depending on which carrier is used for the receiving phone number. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, August 20, 2014 3:03 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI timing settings What about the text portion? Should that never be sent? I was indeed sending the '1', and I will remove that to see if it solves my problem, but I also have the company name in there. I feel like a newb asking such questions, but I've never had this issue before :) Company 1NXXNXX Cheers, j On 08/20/2014 09:46 AM, Eric Wieling wrote: NXXNXX is the correct format of CallerID numbers in NANPA. The leading 1 is not part of any NANPA phone number. Toll free area codes are also not valid for CallerID. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, August 20, 2014 2:41 PM To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI timing settings On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.netmailto:j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don't know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don't know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] PRI timing settings
It's possible that Sprint is burping on the name. Try first dropping the 1. Then try dropping the name also, if necessary. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, August 20, 2014 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI timing settings What about the text portion? Should that never be sent? I was indeed sending the '1', and I will remove that to see if it solves my problem, but I also have the company name in there. I feel like a newb asking such questions, but I've never had this issue before :) Company 1NXXNXX Cheers, j On 08/20/2014 09:46 AM, Eric Wieling wrote: NXXNXX is the correct format of CallerID numbers in NANPA. The leading 1 is not part of any NANPA phone number. Toll free area codes are also not valid for CallerID. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, August 20, 2014 2:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI timing settings On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don't know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don't know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] customizing internal calls
I would let the phone handle the different ring tones, if possible. For my phones a SIPAddHeader with something like Alert-Info: http://127.0.0.1/Ringer3 does the trick, but the syntax might be vendor specific. The other problem should be taken care of with call queues. jg -- _ -- 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 listening on undefined IP as per bindaddr
Hello all, I am running asterisk on VMs with standby heartbeat configuration, Heartbeat assigns a virtual IP 172.20.255.40 on machine afterwards asterisk is started. In the sip.conf, I have explicitly define bindaddr=172.20.255.40 but sometimes I see packets coming from physical IP 172.20.255.41 I have both tcp and udp transport enabled Here is the lsof -ni :5060 output asterisk 2878 asterisk 613r IPv4 40060683 0t0 TCP 172.20.255.41:52381-10.100.210.110:sip (ESTABLISHED) asterisk 2878 asterisk 528u IPv4 29757779 0t0 TCP 172.20.255.41:55627-10.200.14.29:sip (ESTABLISHED) asterisk 2878 asterisk 530u IPv4 19211854 0t0 TCP 172.20.255.40: sip-10.100.157.32:49227 (ESTABLISHED) sip show settings Global Settings: UDP Bindaddress:172.20.255.40:5060 TCP SIP Bindaddress:172.20.255.40:5060 Anyone has idea what could be the reason? Regards, Zohair Raza -- _ -- 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] PRI timing settings
Sadly none of these changes have made any difference. I'll report the resolution for posterity once we find it. Thanks, j On 08/20/2014 10:13 AM, Don Kelly wrote: It’s possible that Sprint is burping on the name. Try first dropping the “1.” Then try dropping the name also, if necessary. --Don *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere *Sent:* Wednesday, August 20, 2014 10:03 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI timing settings What about the text portion? Should that never be sent? I was indeed sending the '1', and I will remove that to see if it solves my problem, but I also have the company name in there. I feel like a newb asking such questions, but I've never had this issue before :) Company 1NXXNXX Cheers, j On 08/20/2014 09:46 AM, Eric Wieling wrote: NXXNXX is the correct format of CallerID numbers in NANPA. The leading 1 is not part of any NANPA phone number. Toll free “area codes” are also not valid for CallerID. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere *Sent:* Wednesday, August 20, 2014 2:41 PM *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI timing settings On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net mailto:j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don’t know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don’t know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] PRI timing settings
PRI intense debug should show all you need to fix this. On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net wrote: Sadly none of these changes have made any difference. I'll report the resolution for posterity once we find it. Thanks, j On 08/20/2014 10:13 AM, Don Kelly wrote: It’s possible that Sprint is burping on the name. Try first dropping the “1.” Then try dropping the name also, if necessary. --Don *From:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere *Sent:* Wednesday, August 20, 2014 10:03 AM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI timing settings What about the text portion? Should that never be sent? I was indeed sending the '1', and I will remove that to see if it solves my problem, but I also have the company name in there. I feel like a newb asking such questions, but I've never had this issue before :) Company 1NXXNXX Cheers, j On 08/20/2014 09:46 AM, Eric Wieling wrote: NXXNXX is the correct format of CallerID numbers in NANPA. The leading 1 is not part of any NANPA phone number. Toll free “area codes” are also not valid for CallerID. *From:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere *Sent:* Wednesday, August 20, 2014 2:41 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI timing settings On 08/20/2014 07:58 AM, Scott L. Lykens wrote: On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote: I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as busy. I don’t know what expectations for CLID your carrier might have, or for that matter the upstream carrier, however, we found through our CLEC here in the US that while the CLEC was happy to take e.164 formatted numbers from us as CLID, Global Crossing would reject them further upstream resulting in our calls to many toll frees being rejected. Switching to 10 digit CLID on all outbound calls through that PRI solved the problem. I don’t know if this is your problem but be sure your CLID is in the most simple format possible for your region to help rule it out. sl This makes me curious... what *is* the simplest format possible for NANPA numbers? I'm sure there must be a spec to conform to. Can anyone point me to it? Cheers, j -- _ -- 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] PRI timing settings
On 8/20/14, 11:28 AM, Steve Totaro wrote: PRI intense debug should show all you need to fix this. Right, the sooner you post this debug here the sooner we can help. Otherwise its just guesswork. On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net mailto:j...@jeff.net wrote: Sadly none of these changes have made any difference. I'll report the resolution for posterity once we find it. Thanks, j -- Technical Support http://www.cellroute.net -- _ -- 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] PRI timing settings
On 08/20/2014 12:04 PM, Andres wrote: On 8/20/14, 11:28 AM, Steve Totaro wrote: PRI intense debug should show all you need to fix this. Right, the sooner you post this debug here the sooner we can help. Otherwise its just guesswork. On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net mailto:j...@jeff.net wrote: Sadly none of these changes have made any difference. I'll report the resolution for posterity once we find it. Thanks, j Ok, here is an intense debug trace. I've replaced the phone numbers to protect the innocent. The smoking gun seems to be this: Ext: 1 Cause: Destination out of order (27) Though I have no idea why... calling the same destination from my cell phone works fine. We only send seven digits for local on-island calls like this, and calls to other carriers work fine with the same format. I'm starting to doubt there is anything I can do to fix this... seems like an issue between my telco and Sprint? Cheers, j astsouth*CLI pri intense debug span 1 Enabled debugging on span 1 PRI Span: 1 t203_expire PRI Span: 1 PRI Span: 1 TEI: 0 State 7(Multi-frame established) PRI Span: 1 V(A)=17, V(S)=17, V(R)=73 PRI Span: 1 K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 PRI Span: 1 T200_id=0, N200=3, T203_id=0 PRI Span: 1 [ 00 01 01 93 ] PRI Span: 1 Supervisory frame: PRI Span: 1 SAPI: 00 C/R: 0 EA: 0 PRI Span: 1 TEI: 000EA: 1 PRI Span: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] PRI Span: 1 N(R): 073 P/F: 1 PRI Span: 1 0 bytes of data PRI Span: 1 -- Starting T200 timer PRI Span: 1 PRI Span: 1 TEI: 0 State 8(Timer recovery) PRI Span: 1 V(A)=17, V(S)=17, V(R)=73 PRI Span: 1 K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 PRI Span: 1 T200_id=8192, N200=3, T203_id=0 PRI Span: 1 [ 02 01 01 23 ] PRI Span: 1 Supervisory frame: PRI Span: 1 SAPI: 00 C/R: 1 EA: 0 PRI Span: 1 TEI: 000EA: 1 PRI Span: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] PRI Span: 1 N(R): 017 P/F: 1 PRI Span: 1 0 bytes of data PRI Span: 1 PRI Span: 1 TEI: 0 State 8(Timer recovery) PRI Span: 1 V(A)=17, V(S)=17, V(R)=73 PRI Span: 1 K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 PRI Span: 1 T200_id=8192, N200=3, T203_id=0 PRI Span: 1 [ 02 01 01 93 ] PRI Span: 1 Supervisory frame: PRI Span: 1 SAPI: 00 C/R: 1 EA: 0 PRI Span: 1 TEI: 000EA: 1 PRI Span: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] PRI Span: 1 N(R): 073 P/F: 1 PRI Span: 1 0 bytes of data PRI Span: 1 -- Got ACK for N(S)=17 to (but not including) N(S)=17 PRI Span: 1 Done handling message for SAPI/TEI=0/0 PRI Span: 1 PRI Span: 1 TEI: 0 State 8(Timer recovery) PRI Span: 1 V(A)=17, V(S)=17, V(R)=73 PRI Span: 1 K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 PRI Span: 1 T200_id=8192, N200=3, T203_id=0 PRI Span: 1 [ 00 01 01 23 ] PRI Span: 1 Supervisory frame: PRI Span: 1 SAPI: 00 C/R: 0 EA: 0 PRI Span: 1 TEI: 000EA: 1 PRI Span: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] PRI Span: 1 N(R): 017 P/F: 1 PRI Span: 1 0 bytes of data PRI Span: 1 -- Got ACK for N(S)=17 to (but not including) N(S)=17 PRI Span: 1 -- Stopping T200 timer PRI Span: 1 -- Starting T203 timer PRI Span: 1 Done handling message for SAPI/TEI=0/0 == Using SIP RTP CoS mark 5 -- Executing [998@business:1] Dial(SIP/bolongo-1c78, DAHDI/g0/998,60) in new stack PRI Span: 1 -- Making new call for cref 32897 -- Requested transfer capability: 0x00 - SPEECH PRI Span: 1 PRI Span: 1 DL-DATA request PRI Span: 1 Protocol Discriminator: Q.931 (8) len=56 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 129/0x81) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 TEI=0 Transmitting N(S)=17, window is open V(A)=17 K=7 PRI Span: 1 PRI Span: 1 TEI: 0 State 7(Multi-frame established) PRI Span: 1 V(A)=17, V(S)=17, V(R)=73 PRI Span: 1 K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 PRI Span: 1 T200_id=0, N200=3, T203_id=8192 PRI Span: 1 [ 00 01 22 92 08 02 00 81 05 04 03 80 90 a2 18 03 a1 83 81 1e 02 80 83 28 0b b1 33 34 30 37 37 35 31 38 30 30 6c 0c 21 80 33 34 30 37 37 35 31 38 30 30 70 08 80 39 39 38 39 39 36 35 ] PRI Span: 1 Informational frame: PRI Span: 1 SAPI: 00 C/R: 0 EA: 0 PRI Span: 1 TEI: 000EA: 1 PRI Span: 1 N(S): 017 0: 0 PRI Span: 1 N(R): 073 P: 0 PRI Span: 1 56 bytes of data PRI Span: 1 Protocol Discriminator: Q.931 (8) len=56 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 129/0x81) (Sent from originator) PRI Span: 1 Message Type: SETUP (5) PRI Span: 1 [04 03 80 90 a2] PRI Span: 1 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 1 User information layer 1: u-Law (34) PRI Span: 1 [18 03 a1 83 81] PRI Span: 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0
Re: [asterisk-users] PRI timing settings
Do you also dial only 7 digits when calling from your cellphone when it works? Have you tried using the whole number in your dial? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, August 20, 2014 5:29 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI timing settings On 08/20/2014 12:04 PM, Andres wrote: Ok, here is an intense debug trace. I've replaced the phone numbers to protect the innocent. The smoking gun seems to be this: Ext: 1 Cause: Destination out of order (27) Though I have no idea why... calling the same destination from my cell phone works fine. We only send seven digits for local on-island calls like this, and calls to other carriers work fine with the same format. I'm starting to doubt there is anything I can do to fix this... seems like an issue between my telco and Sprint? Cheers, -- _ -- 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] PRI timing settings
Ok, here is an intense debug trace. I've replaced the phone numbers to protect the innocent. The smoking gun seems to be this: Ext: 1 Cause: Destination out of order (27) Though I have no idea why... calling the same destination from my cell phone works fine. We only send seven digits for local on-island calls like this, and calls to other carriers work fine with the same format. I'm starting to doubt there is anything I can do to fix this... seems like an issue between my telco and Sprint? Cheers, j [snip long trace] From here http://www.cnes.com/causecodes.html: *Cause No. 27 - destination out of order [Q.850] * This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line. I am betting this is simply what my telco is sending because they were unable to pass the call on to Sprint. It would be more informative perhaps to see what kind of trace the telco has of the handoff to Sprint. I'm going to see if they will give it to me. Cheers, j -- _ -- 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] PRI timing settings
Yes from the local cell phone seven digits is enough. We have tried sending ten digits with the same result over the PRI. Cheers, j On 08/20/2014 12:38 PM, Eric Wieling wrote: Do you also dial only 7 digits when calling from your cellphone when it works? Have you tried using the whole number in your dial? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere *Sent:* Wednesday, August 20, 2014 5:29 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI timing settings On 08/20/2014 12:04 PM, Andres wrote: Ok, here is an intense debug trace. I've replaced the phone numbers to protect the innocent. The smoking gun seems to be this: Ext: 1 Cause: Destination out of order (27) Though I have no idea why... calling the same destination from my cell phone works fine. We only send seven digits for local on-island calls like this, and calls to other carriers work fine with the same format. I'm starting to doubt there is anything I can do to fix this... seems like an issue between my telco and Sprint? Cheers, -- _ -- 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] Dispatching calls question
I have a question about dispatching calls... If I try to dispatch a call on line 1 using the AMI and I check in my table to see if line 1 is available and it is So I have done my checking now I dispatch my call and at that same time a call comes in on line 1 and now its no longer available for me to make a call, I connect on AMI and my call fails How do I prevent this from happening? Sure I can start at 23 instead of 1 and work down instead of up but eventually the same thing may happen. I'm using Asterisk 11.11 if that matters. 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] Dispatching calls question
On Wed, 20 Aug 2014, Jerry Geis wrote: I have a question about dispatching calls... If I try to dispatch a call on line 1 using the AMI and I check in my table to see if line 1 is available and it is So I have done my checking now I dispatch my call and at that same time a call comes in on line 1 and now its no longer available for me to make a call, I connect on AMI and my call fails How do I prevent this from happening? Sure I can start at 23 instead of 1 and work down instead of up but eventually the same thing may happen. If you're using something like MySQL, use 'get_lock/release_lock.' If you're using some other database, see what locking features you have available. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Dispatching calls question
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, August 20, 2014 9:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dispatching calls question On Wed, 20 Aug 2014, Jerry Geis wrote: I have a question about dispatching calls... If I try to dispatch a call on line 1 using the AMI and I check in my table to see if line 1 is available and it is So I have done my checking now I dispatch my call and at that same time a call comes in on line 1 and now its no longer available for me to make a call, I connect on AMI and my call fails How do I prevent this from happening? Sure I can start at 23 instead of 1 and work down instead of up but eventually the same thing may happen. If you're using something like MySQL, use 'get_lock/release_lock.' If you're using some other database, see what locking features you have available. Asterisk 1.8 and later have lock functions available in the dialplan. This might be better if you have a single Asterisk server. pbx*CLI core show functions like LOCK Matching Custom Functions: LOCK LOCK(lockname) Attempt to obtain a named mutex. TRYLOCK TRYLOCK(lockname)Attempt to obtain a named mutex. UNLOCKUNLOCK(lockname) Unlocks a named mutex. 3 matching custom functions installed. -- _ -- 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
