Re: [asterisk-users] looking for some quality testers for zoiper softphone for android.
On 31.5.2012 г. 20:43 ч., Patrick Lists wrote: Hi zoa, On 31-05-12 17:39, joachim wrote: Ellow, We released zoiper for Android today, available for free here: https://play.google.com/store/apps/details?id=com.zoiper.android.app SIP and IAX is supported, should work quite well, unfortunately it is really hard to test all android and hardware combinations. Any android lovers out there to send us some feedback ? Preferably with packet capture skills ? I am mainly looking for feedback on the audio quality, audio delay and if everything looks ok in the gui. Had a quick look on Google Play. Are g729, AMR-NB and SRTP missing in the description or does it not have those features (yet)? Regards, Patrick Hello, We do not have those features yet, we will add them in the future. Joachim -- _ -- 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] Queue callers with Callback option without lose their place
I believe you want your caller to request for a callback while he/she waits in a queue and when your agents are free, you want to call him back and place in a same position in a Queue where he/she has left the Queue. There exists an ugly(!) way of doing this. (1)Set parameter 'context' in queues.conf to some real context available in your dialplan (2)Set 'setqueueentryvar' and 'setqueuevar' to yes in queues.conf (3)Set paramet 'periodic-announce' to a custom audio file name announcing to caller somethink like ..'To get a callback press any key any'.(This sends the caller into context set by 'context' parameter when s/he presses any key while waiting in a queue) (4)A variable 'QUEUEPOSITION' would give you a last position of caller in a queue. (You can get this variable in a context set by 'context' parameter. Store the value somewhere in Database) (5)When you think your Agents are free, Generate a callfile OR use AMI to call the caller who has requested a callback. (6)Once call is answered, send him to Queue application with 'position' parameter set to the value of 'QUEUEPOSITION' of caller from database. --Satish Barot On Thu, May 31, 2012 at 9:18 PM, equis software equissoftw...@gmail.comwrote: Is there any option in Asterisk distribution of this? Thanks. -- _ -- 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] Queue callers with Callback option without lose their place
http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback On Fri, Jun 1, 2012 at 1:45 PM, Satish Barot satish4aster...@gmail.comwrote: I believe you want your caller to request for a callback while he/she waits in a queue and when your agents are free, you want to call him back and place in a same position in a Queue where he/she has left the Queue. There exists an ugly(!) way of doing this. (1)Set parameter 'context' in queues.conf to some real context available in your dialplan (2)Set 'setqueueentryvar' and 'setqueuevar' to yes in queues.conf (3)Set paramet 'periodic-announce' to a custom audio file name announcing to caller somethink like ..'To get a callback press any key any'.(This sends the caller into context set by 'context' parameter when s/he presses any key while waiting in a queue) (4)A variable 'QUEUEPOSITION' would give you a last position of caller in a queue. (You can get this variable in a context set by 'context' parameter. Store the value somewhere in Database) (5)When you think your Agents are free, Generate a callfile OR use AMI to call the caller who has requested a callback. (6)Once call is answered, send him to Queue application with 'position' parameter set to the value of 'QUEUEPOSITION' of caller from database. --Satish Barot On Thu, May 31, 2012 at 9:18 PM, equis software equissoftw...@gmail.comwrote: Is there any option in Asterisk distribution of this? Thanks. -- _ -- 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
[asterisk-users] Fax over IP ?
Hi all, Couple of things I would like ask, does Asterisk provides free license for FoIP (for 1 channel) or need to purchase it? Couple of years back, I was able to send and receive the fax using Digium T1 card, in term of FoIP how can I able to receive fax from traditional telephone lines / T1 lines? As far my understanding, the functionality for FoIP is to send fax to email or receive fax from email i.e. using T.38 protocol. The thing I would like to know how I can implement this solution i.e. receiving fax via IP? Correct me if I'm wrong, while receiving fax from traditional telephone lines will the topology looks like as listed below; PSTN Lines -- Asterisk (mounted a T1/ analog card) -- IP -- Asterisk (receive Fax over IP) or else? Please advice. -- Regards, Ahmed Munir Chohan -- _ -- 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] Fax over IP ?
- Original Message - Hi all, Couple of things I would like ask, does Asterisk provides free license for FoIP (for 1 channel) or need to purchase it? Couple of years back, I was able to send and receive the fax using Digium T1 card, in term of FoIP how can I able to receive fax from traditional telephone lines / T1 lines? As far my understanding, the functionality for FoIP is to send fax to email or receive fax from email i.e. using T.38 protocol. The thing I would like to know how I can implement this solution i.e. receiving fax via IP? Correct me if I'm wrong, while receiving fax from traditional telephone lines will the topology looks like as listed below; PSTN Lines -- Asterisk (mounted a T1/ analog card) -- IP -- Asterisk (receive Fax over IP) or else? FoIP typically means the fax session traverses an IP link at some point, most commonly at the 'last mile'. What happens to the fax after that is up to your requirements. The faxes can be emailed out, stored in a web application, printed to a printer, etc. The possibilities are endless. Asterisk does have a few options for faxing. Those are most notably: 1. Fax for Asterisk - Free license available for 1 channel, or paid licenses for 2+ channels 2. app_fax (I think this is the current module name) - Free fax module for Asterisk, no channel limit, based on SpanDSP 3. Hylafax+ and IAXmodem - Most complicated method of fax setup, but most robust and reliable (in my testing). Would require use of Asterisk 10 with T.38 gateway functionality for proper fax reception. Just keep in mind raw fax audio over VoIP is a bad idea, see here: http://www.soft-switch.org/foip.html If you can provide some additional details on what you're planning to do, we can give more info. --Tim -- _ -- 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] Fax Server for Asterisk
- Original Message - Hi Tim, Unfortunately i can't reproduce the scenario because it was a long time ago. But it would be nice to hear from you, what things can be verified within fax and Asterisk? Any TIP on wireshark monitoring? Within Asterisk, the debug logs can be helpful for routing/connectivity diagnostics. With Hylafax, all of your details will be found in the session logs in /var/spool/hylafax/log. Here you can see each session's interaction with the remote fax device. It is an art deciphering the various protocols, but the folks on the Hylafax lists are incredibly helpful until you've learned the magic of understanding the logs directly. :) --Tim -- _ -- 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] Half-height PCIe analog FXO card
Hi, Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant microserver for $nuppence, which I'd hoped to migrate my Asterisk setup onto. I currently use an A400P analog card, but the ProLiant only has PCIe slots, and they're short ones too, so I can't use an A400E card. Even the Sangoma cards, which seem to be low profile, have full-height brackets on them - which, of course, won't fit in the box. Is it just me, or is this whole half-height PCIe thing a complete b***ocks? Any advice appreciated. I'd prefer not to have to spend mega$ on this, the server only cost $200, it seems silly to spend $1000 on a PCI to PCIe converter (Magma.com) to keep using a $100 card... Cheers, Ade. -- _ -- 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] Half-height PCIe analog FXO card
Last time I checked (a few years ago) Sangoma has half height brackets available. Contact their support or sales. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ade Vickers Sent: Friday, June 01, 2012 10:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Half-height PCIe analog FXO card Hi, Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant microserver for $nuppence, which I'd hoped to migrate my Asterisk setup onto. I currently use an A400P analog card, but the ProLiant only has PCIe slots, and they're short ones too, so I can't use an A400E card. Even the Sangoma cards, which seem to be low profile, have full-height brackets on them - which, of course, won't fit in the box. Is it just me, or is this whole half-height PCIe thing a complete b***ocks? Any advice appreciated. I'd prefer not to have to spend mega$ on this, the server only cost $200, it seems silly to spend $1000 on a PCI to PCIe converter (Magma.com) to keep using a $100 card... Cheers, Ade. -- _ -- 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] Half-height PCIe analog FXO card
Ade- Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant microserver for $nuppence, which I'd hoped to migrate my Asterisk setup onto. I currently use an A400P analog card, but the ProLiant only has PCIe slots, and they're short ones too, so I can't use an A400E card. Even the Sangoma cards, which seem to be low profile, have full-height brackets on them - which, of course, won't fit in the box. Is it just me, or is this whole half-height PCIe thing a complete b***ocks? Hehe, that makes me laugh. Thanks to huge power-hungry GPU boards, the full length/height slots -- which I thought were dead in the late 1990s -- have come back and established their rightful position :-) -Jeff -- _ -- 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] Session-timers and TCP
All, We are having issues with one of our customers. They typically are using remote sip clients on smart phones. For the purpose of allowing the apps to work properly in the background we have to utilize TCP which works fine. The problem comes up when the softphone loses connectivity for some reason. The session timers are not ending the call as they do on a UDP session. Basically from the sip debug it sends the re-invite for the session timer according to the sip debug and it appears all is fine instead of not getting a response back from the client and disconnecting the call as it does with udp. There is no way it is getting a response back from the client however as the client has no network connectivity. I have run some tcpdump's on the server and when tracing the call I actually never see those re-invites going out at all from the server. We are running asterisk 1.8.7.0 currently. I can reproduce the issue at will by establishing a call from a softphone and then putting it into airplane mode to simulate the connectivity loss. Are session-timers expected to work with tcp? If so can anyone tell me where to look to see what might be going on? Thanks in Advance. -- _ -- 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] Fax over IP?
- Original Message - Hi Tim, Thanks for your response. Here is my topology as listing down below; PSTN Line -- Cisco Voice GW -- IP Cloud -- Asterisk Will Asterisk able to receive the fax (as in topology above) using its' fax module? In sip.conf I enabled fax detection and T.38. Actually I don't want to use Hylafax + iaxmodem as per requirement. If your Cisco voice gateway can deliver the calls using T.38, that should give you decent reliability. You'll want to us Asterisk 10 which has the best T.38 support at this point (compared to older releases). The receiving side of the equation then becomes whether to use Fax for Asterisk (commercial, 1 free channel, 2+ paid), or the included SpanDSP based fax module. --Tim -- _ -- 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 10.4.0 GotoIf to label problem when DUNDi active
On Wed, 30 May 2012 18:02:00 + Noah Engelberth n...@directlinkcomputers.com wrote: I have a hotdesking environment at my main office, and up until today, the GotoIf that jumps straight to voicemail if a user isn't log in was working just fine by label. Today, I deployed DUNDi to a satellite office, and now the GotoIf isn't jumping to the right place. If I replace the label with a priority number, it jumps correctly. Alternatively, if I disable the switch statement for DUNDi, it jumps correctly. But with the DUNDi switch in service and the named label to jump to, it gives me this error: [May 30 13:57:24] WARNING[6654]: pbx.c:10747 pbx_parseable_goto: Priority 'not_logged_in' must be a number 0, or valid label What do you get when you run dialplan show internal on the Asterisk CLI? Does it show the not_logged_in label? Have you looked at the CLI output when you reload? There may be a syntax error before your not_logged_in line... Also, along the same vein, you might try moving the [internal-privledged] context (with the switch/DUNDi line) to below the [internal] one. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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 10.4.0 GotoIf to label problem when DUNDi active
- dialplan show internal shows the label (and if I change it, it changes to the right one, etc.) - No syntax errors or warnings showing in the CLI when I reload with verbosity at 3. - I'll check moving things around on Monday, but I find it odd that all I have to do to make the GotoIf work with a label is to comment out the switch (no other changes). Noah -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace Sent: Friday, June 01, 2012 6:42 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 10.4.0 GotoIf to label problem when DUNDi active On Wed, 30 May 2012 18:02:00 + Noah Engelberth n...@directlinkcomputers.com wrote: I have a hotdesking environment at my main office, and up until today, the GotoIf that jumps straight to voicemail if a user isn't log in was working just fine by label. Today, I deployed DUNDi to a satellite office, and now the GotoIf isn't jumping to the right place. If I replace the label with a priority number, it jumps correctly. Alternatively, if I disable the switch statement for DUNDi, it jumps correctly. But with the DUNDi switch in service and the named label to jump to, it gives me this error: [May 30 13:57:24] WARNING[6654]: pbx.c:10747 pbx_parseable_goto: Priority 'not_logged_in' must be a number 0, or valid label What do you get when you run dialplan show internal on the Asterisk CLI? Does it show the not_logged_in label? Have you looked at the CLI output when you reload? There may be a syntax error before your not_logged_in line... Also, along the same vein, you might try moving the [internal-privledged] context (with the switch/DUNDi line) to below the [internal] one. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] Asterisk pickup call on first ring
Thanks Eric for the prompt reply :) Honestly I still need the caller id but I already strugle for around 1-2months to get the caller id work on my system :( yesterday I bought a caller id converter hoping it will solve my problem but look like it's not. I'm still trying to get the caller id to work. So there is no other setting except disable the caller id detection for the system to pickup incoming call at the first ring? Thanks a lot. On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: Try usecallerid=no The immediate= option is mainly for FXS ports and is almost never used. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk pickup call on first ring Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 pickup call on first ring
That depends on what country you are in. In the USA CallerID information is sent between the first and 2nd ring.Asterisk defaults to expecting USA style CallerID. If you are not in the USA then you'll have to research on how to get CallerID working with Asterisk for your country. Search the mailing list archives. Try disabling CallerID in Asterisk just to see if that is actually causing the delay. If it is, you know once you get CallerID working the delay will go away and you won't spend more time trying to fix the ring delay. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk pickup call on first ring Thanks Eric for the prompt reply :) Honestly I still need the caller id but I already strugle for around 1-2months to get the caller id work on my system :( yesterday I bought a caller id converter hoping it will solve my problem but look like it's not. I'm still trying to get the caller id to work. So there is no other setting except disable the caller id detection for the system to pickup incoming call at the first ring? Thanks a lot. On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: Try usecallerid=no The immediate= option is mainly for FXS ports and is almost never used. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk pickup call on first ring Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] Asterisk pickup call on first ring
Ok,understand :) Just now I change usecallerid=yes to usecallerid=no in chan_dahdi.conf and do a module reload chan_dahdi but it still answer the incoming call until 3-4 rings. Do I need to restart the server or module reload is enough since I can't restart the server right now. Thanks :) On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: That depends on what country you are in. In the USA CallerID information is sent between the first and 2nd ring.Asterisk defaults to expecting USA style CallerID. If you are not in the USA then you'll have to research on how to get CallerID working with Asterisk for your country. Search the mailing list archives. Try disabling CallerID in Asterisk just to see if that is actually causing the delay. If it is, you know once you get CallerID working the delay will go away and you won't spend more time trying to fix the ring delay. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk pickup call on first ring Thanks Eric for the prompt reply :) Honestly I still need the caller id but I already strugle for around 1-2months to get the caller id work on my system :( yesterday I bought a caller id converter hoping it will solve my problem but look like it's not. I'm still trying to get the caller id to work. So there is no other setting except disable the caller id detection for the system to pickup incoming call at the first ring? Thanks a lot. On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: Try usecallerid=no The immediate= option is mainly for FXS ports and is almost never used. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk pickup call on first ring Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 pickup call on first ring
Any changes inside chan_dahdi requires asterisk restart. you can restart asterisk gracefully, where by asterisk will honor the existing calls, but wont honor new calls till it restarts. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Sat, Jun 2, 2012 at 10:20 AM, Satria Anamarta anam.satri...@gmail.comwrote: Ok,understand :) Just now I change usecallerid=yes to usecallerid=no in chan_dahdi.conf and do a module reload chan_dahdi but it still answer the incoming call until 3-4 rings. Do I need to restart the server or module reload is enough since I can't restart the server right now. Thanks :) On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: That depends on what country you are in. In the USA CallerID information is sent between the first and 2nd ring.Asterisk defaults to expecting USA style CallerID. If you are not in the USA then you'll have to research on how to get CallerID working with Asterisk for your country. Search the mailing list archives. Try disabling CallerID in Asterisk just to see if that is actually causing the delay. If it is, you know once you get CallerID working the delay will go away and you won't spend more time trying to fix the ring delay. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk pickup call on first ring Thanks Eric for the prompt reply :) Honestly I still need the caller id but I already strugle for around 1-2months to get the caller id work on my system :( yesterday I bought a caller id converter hoping it will solve my problem but look like it's not. I'm still trying to get the caller id to work. So there is no other setting except disable the caller id detection for the system to pickup incoming call at the first ring? Thanks a lot. On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: Try usecallerid=no The immediate= option is mainly for FXS ports and is almost never used. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk pickup call on first ring Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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
Re: [asterisk-users] Asterisk pickup call on first ring
This is incorrect. The vast majority of settings in chan_dahdi.conf are applied when you do a module reload chan_dahdi.so You cannot change signaling, switchtype, or add or remove channels (I'm sure there are a few others) on a module reload, but most settings will be applied on a reload. If you have no active dahdi channels you can usually you can do a module unload chan_dahdi.so and a module load chan_dahdi.so to totally reload dahdi without restarting your PBX. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani Sent: Saturday, June 02, 2012 12:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk pickup call on first ring Any changes inside chan_dahdi requires asterisk restart. you can restart asterisk gracefully, where by asterisk will honor the existing calls, but wont honor new calls till it restarts. -- _ -- 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 pickup call on first ring
Thanks, Mitul :) What if we change the chan_extra.conf do we also need to restart the server, can not only module reload chan_extra? Thanks again :) On 6/2/12, Mitul Limbani mi...@enterux.in wrote: Any changes inside chan_dahdi requires asterisk restart. you can restart asterisk gracefully, where by asterisk will honor the existing calls, but wont honor new calls till it restarts. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Sat, Jun 2, 2012 at 10:20 AM, Satria Anamarta anam.satri...@gmail.comwrote: Ok,understand :) Just now I change usecallerid=yes to usecallerid=no in chan_dahdi.conf and do a module reload chan_dahdi but it still answer the incoming call until 3-4 rings. Do I need to restart the server or module reload is enough since I can't restart the server right now. Thanks :) On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: That depends on what country you are in. In the USA CallerID information is sent between the first and 2nd ring.Asterisk defaults to expecting USA style CallerID. If you are not in the USA then you'll have to research on how to get CallerID working with Asterisk for your country. Search the mailing list archives. Try disabling CallerID in Asterisk just to see if that is actually causing the delay. If it is, you know once you get CallerID working the delay will go away and you won't spend more time trying to fix the ring delay. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk pickup call on first ring Thanks Eric for the prompt reply :) Honestly I still need the caller id but I already strugle for around 1-2months to get the caller id work on my system :( yesterday I bought a caller id converter hoping it will solve my problem but look like it's not. I'm still trying to get the caller id to work. So there is no other setting except disable the caller id detection for the system to pickup incoming call at the first ring? Thanks a lot. On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: Try usecallerid=no The immediate= option is mainly for FXS ports and is almost never used. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk pickup call on first ring Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device --
Re: [asterisk-users] Asterisk pickup call on first ring
I hope you understand restart as restarting asterisk service. restart Asterisk (service asterisk restart) or from CLI - restart gracefully now (relevant command line) and not rebooting the server. chan_xtra also utilizes similar hooks for the GSM cards. module unloading and loading is also a good idea. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Sat, Jun 2, 2012 at 10:44 AM, Satria Anamarta anam.satri...@gmail.comwrote: Thanks, Mitul :) What if we change the chan_extra.conf do we also need to restart the server, can not only module reload chan_extra? Thanks again :) On 6/2/12, Mitul Limbani mi...@enterux.in wrote: Any changes inside chan_dahdi requires asterisk restart. you can restart asterisk gracefully, where by asterisk will honor the existing calls, but wont honor new calls till it restarts. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Sat, Jun 2, 2012 at 10:20 AM, Satria Anamarta anam.satri...@gmail.comwrote: Ok,understand :) Just now I change usecallerid=yes to usecallerid=no in chan_dahdi.conf and do a module reload chan_dahdi but it still answer the incoming call until 3-4 rings. Do I need to restart the server or module reload is enough since I can't restart the server right now. Thanks :) On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: That depends on what country you are in. In the USA CallerID information is sent between the first and 2nd ring.Asterisk defaults to expecting USA style CallerID. If you are not in the USA then you'll have to research on how to get CallerID working with Asterisk for your country. Search the mailing list archives. Try disabling CallerID in Asterisk just to see if that is actually causing the delay. If it is, you know once you get CallerID working the delay will go away and you won't spend more time trying to fix the ring delay. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk pickup call on first ring Thanks Eric for the prompt reply :) Honestly I still need the caller id but I already strugle for around 1-2months to get the caller id work on my system :( yesterday I bought a caller id converter hoping it will solve my problem but look like it's not. I'm still trying to get the caller id to work. So there is no other setting except disable the caller id detection for the system to pickup incoming call at the first ring? Thanks a lot. On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: Try usecallerid=no The immediate= option is mainly for FXS ports and is almost never used. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Saturday, June 02, 2012 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk pickup call on first ring Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- Bandwidth and
Re: [asterisk-users] Asterisk pickup call on first ring
Hi Eric, By saying signalling do you also mean a caller id signalling? Thanks :) On 6/2/12, Eric Wieling ewiel...@nyigc.com wrote: This is incorrect. The vast majority of settings in chan_dahdi.conf are applied when you do a module reload chan_dahdi.so You cannot change signaling, switchtype, or add or remove channels (I'm sure there are a few others) on a module reload, but most settings will be applied on a reload. If you have no active dahdi channels you can usually you can do a module unload chan_dahdi.so and a module load chan_dahdi.so to totally reload dahdi without restarting your PBX. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani Sent: Saturday, June 02, 2012 12:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk pickup call on first ring Any changes inside chan_dahdi requires asterisk restart. you can restart asterisk gracefully, where by asterisk will honor the existing calls, but wont honor new calls till it restarts. -- _ -- 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 -- Sent from my mobile device -- _ -- 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
