[asterisk-users] Identify port on Khomp card.
Greetings. I've plugged 3 analog lines on an ethernet cable in an Khomp card to receive it's incoming calls. Without any configuration, when I call those numbers the asterisk server automatically answer the call and play the default music. The problem is: I need to discern the lines and redirect each one to his respective extension. Since they doesn't got any Caller ID Service the asterisk cannot distinguish them and give the default treatment. How do I identify the originating port of the call, the originating line or somehow discern each line to make custom configuration for them? My environment: - Elastix v2.3 - Khomp KFXO IP - All the lines are attached to the card on the ethernet interface. -- Att.* *** Luis H. Forchesatto -- _ -- 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 does not persist callgroup and pickgroup configuration.
Greetings. I'm running asterisk here (elastix) and I have a few extensions configured in it. I have here two different callgroup/pickgroup where the extensions are configured in, but it doesn't work when I try do pickup a call. Looking the config file (sip_additional.conf) I see they are not configured with callgroup/pickgroup, the fields are empty. Manually inserting callgroup/pickgroup on the extensions worked just fine but the next day the configuration just vanished and the extensions was not working. Has someone a clue of whats going on here? -- Att.* *** Luis H. Forchesatto -- _ -- 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] Extension cant pickup calls but can transfer.
Yes, it worked :D Thankyou guys for the help. 2013/3/8 Luis H. Forchesatto > I think I found the problem. Better looking the sip_additional.conf file I > noticed that a few extensions didnt had a callgroup and pickgroup > configured, even with the interface appointing otherwise. > > I manually configured this options and reloader asterisk and now I'm gonna > test the extensions and see if it works now. > > I'll be back with the result soon. > > > > 2013/3/8 A J Stiles > >> On Thursday 07 March 2013, Luis H. Forchesatto wrote: >> > Greetings. >> > >> > I got an extension on my Elastix who cannot pick calls on the other >> > extensions, but It can transfer his calls to the other extensions. When >> > this extension tries to pickup a call pressing *8 it simply does not >> pick >> > it up. Transfering calls works just fine so dtmf may be not the problem. >> > >> > Where should I look? >> >> /etc/asterisk/sip.conf (if it's s SIP phone); otherwise the corresponding >> configuration file for whatever technology it is using. Make sure that >> the >> "pickupgroup" for that extension is the same as the other extensions. >> Then >> $ sudo asterisk -x 'reload' (or enter "reload" in Asterisk CLI) to >> apply the >> change. >> >> -- >> AJS >> >> Answers come *after* questions. >> >> -- >> _ >> -- 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 >> > > > > -- > Att.* > *** > Luis H. Forchesatto > Mail: luis_forchesa...@hotmail.com > -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Extension cant pickup calls but can transfer.
I think I found the problem. Better looking the sip_additional.conf file I noticed that a few extensions didnt had a callgroup and pickgroup configured, even with the interface appointing otherwise. I manually configured this options and reloader asterisk and now I'm gonna test the extensions and see if it works now. I'll be back with the result soon. 2013/3/8 A J Stiles > On Thursday 07 March 2013, Luis H. Forchesatto wrote: > > Greetings. > > > > I got an extension on my Elastix who cannot pick calls on the other > > extensions, but It can transfer his calls to the other extensions. When > > this extension tries to pickup a call pressing *8 it simply does not > pick > > it up. Transfering calls works just fine so dtmf may be not the problem. > > > > Where should I look? > > /etc/asterisk/sip.conf (if it's s SIP phone); otherwise the corresponding > configuration file for whatever technology it is using. Make sure that the > "pickupgroup" for that extension is the same as the other extensions. Then > $ sudo asterisk -x 'reload' (or enter "reload" in Asterisk CLI) to apply > the > change. > > -- > AJS > > Answers come *after* questions. > > -- > _ > -- 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 > -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Extension cant pickup calls but can transfer.
Yes, both are configured in the same ata (linksys pap2) and the configuration options are the same. Call group and pick group are the same for both too. 2013/3/7 Yves A. > is it the same type and make of phone than one of the working ones? > -> compare (dtmf) settings, firmware release etc. > > -> check call-group and pickup group... is the non working extension > configured there? > > regards, > yves > > Am 07.03.2013 20:28, schrieb Luis H. Forchesatto: > > Its only ONE phone who doesnt pickup calls. > > 2013/3/7 Yves A. > >> do you have only ONE phone, that can´t pickup, or is this a general >> problem? >> is pickup configured (feature.conf) AND enabled ? >> >> regards, >> yves >> >> >> Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: >> >> Greetings. >> >> I got an extension on my Elastix who cannot pick calls on the other >> extensions, but It can transfer his calls to the other extensions. When >> this extension tries to pickup a call pressing *8 it simply does not pick >> it up. Transfering calls works just fine so dtmf may be not the problem. >> >> Where should I look? >> >> Any further information needed just ask. >> >> -- >> Att.* >> * >> >> >> >> -- >> _ >> -- 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 >> > > > > -- > Att.* > * > Luis H. Forchesatto > Mail: luis_forchesa...@hotmail.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 > > > > -- > _________ > -- 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 > -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Extension cant pickup calls but can transfer.
Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. > do you have only ONE phone, that can´t pickup, or is this a general > problem? > is pickup configured (feature.conf) AND enabled ? > > regards, > yves > > > Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: > > Greetings. > > I got an extension on my Elastix who cannot pick calls on the other > extensions, but It can transfer his calls to the other extensions. When > this extension tries to pickup a call pressing *8 it simply does not pick > it up. Transfering calls works just fine so dtmf may be not the problem. > > Where should I look? > > Any further information needed just ask. > > -- > Att.* > * > > > > -- > _ > -- 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 > -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Extension cant pickup calls but can transfer.
Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -- _ -- 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] Redirect incoming call to SIP trunk.
Solved. 2013/3/5 Luis H. Forchesatto > Greetings. > > I got two asterisk servers, one is connected to another via sip trunk. The > incoming calls are routed to the time period an if matches is transfered to > the designed extension. If don't, is redirected to a second time period. > Then, if the call matches the second time period it need to be transfered > to the trunk that directs to the second server. > > How do I do to configure it this way? > > The trunk it must be transfered has a outbound route configured too. > > Any further details just ask. > > -- > Att.* > *** > > -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Redirect incoming call to SIP trunk.
Greetings. I got two asterisk servers, one is connected to another via sip trunk. The incoming calls are routed to the time period an if matches is transfered to the designed extension. If don't, is redirected to a second time period. Then, if the call matches the second time period it need to be transfered to the trunk that directs to the second server. How do I do to configure it this way? The trunk it must be transfered has a outbound route configured too. Any further details just ask. -- Att.* *** -- _ -- 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] RJ11 x RJ45
Saudações. Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o conector RJ45 está crimpada com a sequencia 568B e vai ser conectada na placa Khomp, mas a ponta RJ11 eu não sei como deve ficar. Li alguns manuais na internet mas não entendi ao certo como tem que ser feito. -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Monitor extensions status.
Greetings. I got two extensions on my asterisk that autenticates from outside our network, via internet. Is there a way to monitor, in certain time periods, if they are available (online) and send some sort of notification if they don't? There are two extensions to monitor, they belong to the same queue. Both must be available to receive calls at the same time and if one or both are offline I must be notified. They stand behind NAT so making Nagios monitor will either report wrong extension status (monitoring the NATing server/router) or simply useless (unless there's a plugin to monitor asterisk extensions). But anyway...I'll be open to opinions. My environment: - Asterisk 1.6.2.13 - Server running Elastix 2.0.0 - DAHDI v. 2.3.0.1 -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Call hangs when selected queue number 1.
Greetings. I have an asterisk server running and got a few queues configured. When a call comes to the server it redirects to the default IVR who offers the caller the options of which department they want to talk with, starting with the queue 1 (most selected). When the caller press 1 to be redirected the call simply hangs. When other queues are selected the call is transfered and the extensions are called. I solved this problem creating another queue with the extensions of the queue 1 and configured to be redirected to this queue. This problem started do occur with no cause. Before this happens, the queue 1 worked fine for months. -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Volume issue.
Hi experts. Recently I've insalled a PCI Khomp Pane on my server and inserted 4 chips to make call with it. The calls are good and no issue was noticed but I got reports that when someone call the chips the call volume is uncommonly low for both sides and they deploy some failures on the audio, only when the call comes from outside. When an extension at the same network makes a call that goes through the pane the calls are really good. The Khomp pane is an KGSM 40spx with 4 modules. All the 4 modules have chip installed and working. I got outbound routes to make cellphone calls goes out through the chips. If someone calls the chips, the call are redirected to our IVR. I'm using Asterisk 1.6.2.13, dahdi driver version 2.3.0.1. If any information make itself needed just tell me and I post. And sorry for the text, english is not my native language. -- Att.* *** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Issue.
Up? 2012/8/20 Luis H. Forchesatto > Thanks for your answer. > > The logs where posted at pastebin, here the links: > > - Working Phone: http://pastebin.com/q3pHcwna > - Not working phone: http://pastebin.com/iiCHPMmn > > > 2012/8/20 Rusty Newton > >> On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: >> >>> Hi >>> >>> I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of >>> ATA on the network who autenticate the phones: Linksys PAP2, Overtek >>> OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at >>> the same network all with g729 codecs and rfc2833 for the DTMF. Making >>> calls via the Overtek ATA the DTMF works fine but at the others ATA it >>> doesn't. >>> >>> My config: >>> >>> - asterisk 1.6.2.13 >>> - dahdi 2.3.0.1 >>> - The phones connected are all physical phones >>> >> There is additional data you can provide to make it easier for others to >> help out: >> If you can pastebin an Asterisk log including all message types plus >> VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that would >> be very helpful. >> A step beyond that is to also provide a SIP and RTP packet trace so that >> whoever wants to help can look through it in Wireshark. >> >> If you can get the packet trace for the same calls you gather log data >> for, that would be best. >> >> Thanks! >> >> [1] https://wiki.asterisk.org/**wiki/display/AST/Collecting+** >> Debug+Information<https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information> >> >> -- >> Rusty Newton >> Digium, Inc | Open Source Community Support Manager >> 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >> > > > > -- > Att.* > *** > -- Att.* *** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Issue.
Thanks for your answer. The logs where posted at pastebin, here the links: - Working Phone: http://pastebin.com/q3pHcwna - Not working phone: http://pastebin.com/iiCHPMmn 2012/8/20 Rusty Newton > On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: > >> Hi >> >> I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of >> ATA on the network who autenticate the phones: Linksys PAP2, Overtek >> OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at >> the same network all with g729 codecs and rfc2833 for the DTMF. Making >> calls via the Overtek ATA the DTMF works fine but at the others ATA it >> doesn't. >> >> My config: >> >> - asterisk 1.6.2.13 >> - dahdi 2.3.0.1 >> - The phones connected are all physical phones >> > There is additional data you can provide to make it easier for others to > help out: > If you can pastebin an Asterisk log including all message types plus > VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that would > be very helpful. > A step beyond that is to also provide a SIP and RTP packet trace so that > whoever wants to help can look through it in Wireshark. > > If you can get the packet trace for the same calls you gather log data > for, that would be best. > > Thanks! > > [1] https://wiki.asterisk.org/**wiki/display/AST/Collecting+** > Debug+Information<https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information> > > -- > Rusty Newton > Digium, Inc | Open Source Community Support Manager > 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<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- Att.* *** -- _ -- 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] DTMF Issue.
Hi I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA on the network who autenticate the phones: Linksys PAP2, Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at the same network all with g729 codecs and rfc2833 for the DTMF. Making calls via the Overtek ATA the DTMF works fine but at the others ATA it doesn't. My config: - asterisk 1.6.2.13 - dahdi 2.3.0.1 - The phones connected are all physical phones -- Att.* *** -- _ -- 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] Extensions DTMF
I've swapper the fones and the "good" phone stopped working. The "good" device is a Overtek OT-ATA200SP and the "bad" phone device is a Linksys PAP2. 2012/8/18 Luis H. Forchesatto > Hi > > Before I swap the phones, I was wondering if asterisk couldn't be lost > somewhere. I've beem reccomended to restart the server or restart the > asterisk to fix this issue but I'm not sure if this will solve the issue. > > > 2012/8/15 Danny Nicholas > >> On my client box that uses OBI110’s, I write the DTMF traffic out to a >> log. I think you have some sort of setting that is garbling your DTMF >> tones. What happens if you move a “good” phone to a “bad” port? >> >> ** ** >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. >> Forchesatto >> *Sent:* Wednesday, August 15, 2012 11:45 AM >> >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] Extensions DTMF >> >> ** ** >> >> Any clues? >> >> 2012/8/15 Luis H. Forchesatto >> >> 2.3.0.1 >> >> ** ** >> >> 2012/8/15 Danny Nicholas >> >> DAHDI version? >> >> >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. >> Forchesatto >> *Sent:* Wednesday, August 15, 2012 8:49 AM >> >> >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> >> *Subject:* Re: [asterisk-users] Extensions DTMF >> >> >> >> They are all physical phones. They are connected to ATA devices which >> autenticate the server at the local network. The server runs Asterisk >> 1.6.2.13.**** >> >> >> >> Att. >> >> 2012/8/15 Danny Nicholas >> >> More details? What type of phones are on the working and failing >> extensions? What flavor of Asterisk did your Elastix install? >> >> >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. >> Forchesatto >> *Sent:* Wednesday, August 15, 2012 8:43 AM >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* [asterisk-users] Extensions DTMF >> >> >> >> Greetings >> >> >> >> Recently I've noticed some of the extensions on our VoIP server are not >> beign recognized by the IVR of a few destinys I've tested. I press que IVR >> number but it simply don't transfer. This is not ocurring to all >> extensions. I'm using rfc2833 to all extensions and Elastix on CentOS 5.5. >> >> >> >> >> -- >> Att. >> >> >> -- >> _ >> -- 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 >> >> >> >> >> >> >> >> -- >> Att. >> >> >> -- >> _ >> -- 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 >> >> >> >> >> >> ** ** >> >> -- >> Att. >> >> >> >> >> >> ** ** >> >> -- >> Att. >> >> -- >> _ >> -- 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 >> > > > > -- > Att.* > *** > -- Att.* *** -- _ -- 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] Extensions DTMF
Hi Before I swap the phones, I was wondering if asterisk couldn't be lost somewhere. I've beem reccomended to restart the server or restart the asterisk to fix this issue but I'm not sure if this will solve the issue. 2012/8/15 Danny Nicholas > On my client box that uses OBI110’s, I write the DTMF traffic out to a > log. I think you have some sort of setting that is garbling your DTMF > tones. What happens if you move a “good” phone to a “bad” port? > > ** ** > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. > Forchesatto > *Sent:* Wednesday, August 15, 2012 11:45 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Extensions DTMF**** > > ** ** > > Any clues? > > 2012/8/15 Luis H. Forchesatto > > 2.3.0.1 > > ** ** > > 2012/8/15 Danny Nicholas > > DAHDI version? > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. > Forchesatto > *Sent:* Wednesday, August 15, 2012 8:49 AM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* Re: [asterisk-users] Extensions DTMF > > > > They are all physical phones. They are connected to ATA devices which > autenticate the server at the local network. The server runs Asterisk > 1.6.2.13. > > > > Att. > > 2012/8/15 Danny Nicholas > > More details? What type of phones are on the working and failing > extensions? What flavor of Asterisk did your Elastix install? > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. > Forchesatto > *Sent:* Wednesday, August 15, 2012 8:43 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Extensions DTMF > > > > Greetings > > > > Recently I've noticed some of the extensions on our VoIP server are not > beign recognized by the IVR of a few destinys I've tested. I press que IVR > number but it simply don't transfer. This is not ocurring to all > extensions. I'm using rfc2833 to all extensions and Elastix on CentOS 5.5. > > > > > -- > Att. > > > -- > _ > -- 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 > > > > > > > > -- > Att. > > > -- > _ > -- 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 > > > > > > ** ** > > -- > Att. > > > > > > ** ** > > -- > Att. > > -- > _ > -- 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 > -- Att.* *** -- _ -- 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] Extensions DTMF
Any clues? 2012/8/15 Luis H. Forchesatto > 2.3.0.1 > > > 2012/8/15 Danny Nicholas > >> DAHDI version? >> >> ** ** >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. >> Forchesatto >> *Sent:* Wednesday, August 15, 2012 8:49 AM >> >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] Extensions DTMF >> >> ** ** >> >> They are all physical phones. They are connected to ATA devices which >> autenticate the server at the local network. The server runs Asterisk >> 1.6.2.13. >> >> ** ** >> >> Att. >> >> 2012/8/15 Danny Nicholas >> >> More details? What type of phones are on the working and failing >> extensions? What flavor of Asterisk did your Elastix install? >> >> >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. >> Forchesatto >> *Sent:* Wednesday, August 15, 2012 8:43 AM >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* [asterisk-users] Extensions DTMF >> >> >> >> Greetings >> >> >> >> Recently I've noticed some of the extensions on our VoIP server are not >> beign recognized by the IVR of a few destinys I've tested. I press que IVR >> number but it simply don't transfer. This is not ocurring to all >> extensions. I'm using rfc2833 to all extensions and Elastix on CentOS 5.5. >> >> >> >> >> -- >> Att. >> >> >> -- >> _ >> -- 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 >> >> >> >> >> >> ** ** >> >> -- >> Att. >> >> -- >> _ >> -- 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 >> > > > > -- > Att.* > *** > -- Att.* *** -- _ -- 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] Extensions DTMF
2.3.0.1 2012/8/15 Danny Nicholas > DAHDI version? > > ** ** > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. > Forchesatto > *Sent:* Wednesday, August 15, 2012 8:49 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Extensions DTMF > > ** ** > > They are all physical phones. They are connected to ATA devices which > autenticate the server at the local network. The server runs Asterisk > 1.6.2.13. > > ** ** > > Att. > > 2012/8/15 Danny Nicholas > > More details? What type of phones are on the working and failing > extensions? What flavor of Asterisk did your Elastix install? > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. > Forchesatto > *Sent:* Wednesday, August 15, 2012 8:43 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Extensions DTMF > > > > Greetings > > > > Recently I've noticed some of the extensions on our VoIP server are not > beign recognized by the IVR of a few destinys I've tested. I press que IVR > number but it simply don't transfer. This is not ocurring to all > extensions. I'm using rfc2833 to all extensions and Elastix on CentOS 5.5. > > > > > -- > Att. > > > -- > _ > -- 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 > > > > > > ** ** > > -- > Att. > > -- > _ > -- 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 > -- Att.* *** -- _ -- 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] Extensions DTMF
They are all physical phones. They are connected to ATA devices which autenticate the server at the local network. The server runs Asterisk 1.6.2.13. Att. 2012/8/15 Danny Nicholas > More details? What type of phones are on the working and failing > extensions? What flavor of Asterisk did your Elastix install? > > ** ** > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Luis H. > Forchesatto > *Sent:* Wednesday, August 15, 2012 8:43 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Extensions DTMF > > ** ** > > Greetings > > ** ** > > Recently I've noticed some of the extensions on our VoIP server are not > beign recognized by the IVR of a few destinys I've tested. I press que IVR > number but it simply don't transfer. This is not ocurring to all > extensions. I'm using rfc2833 to all extensions and Elastix on CentOS 5.5. > > > ** ** > > -- > Att. > > -- > _ > -- 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 > -- Att.* *** -- _ -- 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] Extensions DTMF
Greetings Recently I've noticed some of the extensions on our VoIP server are not beign recognized by the IVR of a few destinys I've tested. I press que IVR number but it simply don't transfer. This is not ocurring to all extensions. I'm using rfc2833 to all extensions and Elastix on CentOS 5.5. -- Att.* *** -- _ -- 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] Voicemail full.
Yep, maxmsg is set to some value and it reached. To make things work again i've moved que messages to a new directory and voicemail is working now. 2012/8/3 Steve Edwards > Un-top-posting... > > > On Fri, 3 Aug 2012, Luis H. Forchesatto wrote: > > I've made a call to our elastix server and the call was redirected to the >> voicemail, which the user should leave a message. Intead recording the call >> the service returned a message like "Sorry, but the user's mailbox can't >> accept more messages". I'm a little lost in the configs here, what >> parameter should I edit to increase the mailbox capacity or what steps I >> take to 'clean' the mailbox? >> > > On Fri, 3 Aug 2012, Danny Nicholas wrote: > > Looking at my voicemail.conf I note this snippet: >> >> ; Maximum number of messages per folder. If not specified, a default value >> >> ; (100) is used. Maximum value for this option is . >> >> ;maxmsg=100 >> >> So in my case max messages is . >> > > (After a quick glance at the source...) > > If not specified, the limit would be MAXMSG (100) not MAXMSGLIMIT (). > > -- > 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<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- Att.* *** -- _ -- 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] Voicemail full.
Hi I've made a call to our elastix server and the call was redirected to the voicemail, which the user should leave a message. Intead recording the call the service returned a message like "Sorry, but the user's mailbox can't accept more messages". I'm a little lost in the configs here, what parameter should I edit to increase the mailbox capacity or what steps I take to 'clean' the mailbox? -- Att.* *** -- _ -- 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] URA
Hi Recently our asterisk system stopped beign recognized by URA in others telephones exchanges. What's the troubleshoot steps for this issue? -- Att.* *** -- _ -- 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
