Re: [asterisk-users] how asterisk work with VoIP trunk?
Hi Steve, Thanks for reply. Is this method will follow on DID incoming calls too? I mean when we call on DID then call will come to my server and then I want to move this call to any SIP extension. But call will not come to extension just got message *"device not in use". *But device already registered into asterisk server. But thanks you clear my concept into Voip Call routing too. On Thu, Jun 9, 2011 at 12:15 AM, Steve Edwards wrote: > On Wed, 8 Jun 2011, virendra bhati wrote: > > I have working experience of asterisk with PRI lines. Recently I have took >> VoIP routes from my provider. My basic issue is that now how asterisk will >> behave in such case. I mean in PRI call will come as below process >> >> PRI - -> Digium Card - -> Dadhi/Zap - -> Extensions.conf >> >> What will be the VoIP calling call flow in Incoming and outgoing calls? >> > > Eth[x] -> sip.conf -> extensions.conf > > -- > 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 > -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] No IVR listen at device end......SIP phone is working fine
Hi Virendra, It may be problem for rtp packet port forwarding if u can dial through DID number. You need to open rtp port range in firewal. e.g. 10,000 to 20,000 port. please, write how can you dial call mobile or other devices. e.g. DID number, PRI number etc. -- Best Regards, Rajnikant Vanza Call : +91-9737456583 Software Engineer --- Working On Linux,C/C++,Asterisk Technology Gandhinagar - Gujarat On Thu, Jun 9, 2011 at 12:13 AM, virendra bhati wrote: > Hi List, > > When we make calls into asterisk with the help of our mobile, landline > number, Cisco 79XX series then we didn't able to here any IVR which is > playing into asterisk server. But when we dial from SIP softphone then all > is working fine and we are able to here the IVR sound files. > > What is the problem in this case please help me.. > > -- > > > > - > Thanks and regards > > Virendra Bhati > +91-9172341457 > Asterisk Engineer > > > -- > _ > -- 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] How asterisk use pri channel
I hope my understanding is not wrong! (1) DAHDI/i2/25/XXX, is not a valid format for Dial. Rather it should be DAHDI/i2/XXX and it would use a channel from span 2 (/etc/dahdi/system.conf) for outgoing call. (2) To dial from channel 25 , use DAHDI/25/XXX [SATISH] On Thu, Jun 9, 2011 at 9:39 AM, satish patel wrote: > Awesome!! > > Do you know if i want to use only specific channel for call out then how do > i write dialplan ? I want to use channel 25 specific for my extension > > DAHDI/25/ or DAHDI/i2/25/XXX > > -- _ -- 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] How asterisk use pri channel
Awesome!! Do you know if i want to use only specific channel for call out then how do i write dialplan ? I want to use channel 25 specific for my extension DAHDI/25/ or DAHDI/i2/25/XXX > Date: Wed, 8 Jun 2011 17:25:44 -0500 > From: rmudg...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] How asterisk use pri channel > > > We have two pri line and I want to see how asterisk distribute > > outgoing call per channels > > > > I meant it use first last channel 47 or it will use first channel? > > > > Or it will allocate dynamically ? > > Extracted from chan_dahdi.c: > > Dial(DAHDI/pseudo[/extension[/options]]) > Dial(DAHDI/[c|r|d][/extension[/options]]) > Dial(DAHDI/![c|r|d][/extension[/options]]) > Dial(DAHDI/i[/extension[/options]]) > Dial(DAHDI/[i-](g|G|r|R)[c|r|d][/extension[/options]]) > > i - ISDN span channel restriction. > Used by CC to ensure that the CC recall goes out the same span. > Also to make ISDN channel names dialable when the sequence number > is stripped off. (Used by DTMF attended transfer feature.) > > g - channel group allocation search forward > G - channel group allocation search backward > r - channel group allocation round robin search forward > R - channel group allocation round robin search backward > > c - Wait for DTMF digit to confirm answer > r - Set distintive ring cadance number > d - Force bearer capability for ISDN/SS7 call to digital. > > All are valid for v1.8 and trunk. The i option and ! option > are not valid earlier than v1.8. > > Richard > > -- > _ > -- 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] Change to pickups in Asterisk 1.8 - not working on local channels?
Hello all,
We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups
which previously worked fine have stopped working.
Can anyone advise if there has been a change in how pickups work?
Here is an example where 1000101 is trying to pick up a call to 1000103:
AGI Rx << EXEC Dial
"Local/1000103@product-pickup
/n,60,M(product-answered^0^1306286740.11)orL(360:6)"
-- AGI Script Executing Application: (Dial) Options:
(Local/1000103@product-pickup
/n,60,M(product-answered^0^1306286740.11)orL(360:6))
> Limit Data for this call:
> timelimit = 360 ms (3600.000 s)
> play_warning = 6 ms (60.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 0 ms (0.000 s)
> start_sound=
> warning_sound = timeleft
> end_sound =
-- Called 1000103@product-pickup/n
-- Executing [1000103@product-pickup:1]
Pickup("Local/1000103@product-pickup-db70;2", "1000103@product-phone") in
new stack
[May 25 11:25:40] NOTICE[1020]: app_directed_pickup.c:313 pickup_exec: No
target channel found for 1000103.
-- Auto fallthrough, channel 'Local/1000103@product-pickup-db70;2'
status is 'UNKNOWN'
The context doing the pickup looks like:
[product-pickup]
exten => _[0-9*#]!, 1, Pickup(${EXTEN}@product-phone)
Thanks for any advice,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
--
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Re: [asterisk-users] After wiki.asterisk.org was upgraded my user no loger exists.
Do you mind checking again? I'm now able to access my account again. Yes, everything is Ok. now, even my documents on personal space. -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- 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] issues.asterisk.org/jira not working
If I click on the link below, without jira, Safari goes to here: https://issues.asterisk.org/main_page.php And yes it works. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 8, 2011, at 1:54 PM, Kevin P. Fleming wrote: > On 06/08/2011 02:27 PM, Andrew Latham wrote: >> On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant wrote: >>> A number of people are reporting that Safari is not working properly with >>> JIRA. Use Firefox or Chrome for now. >>> >>> -- >>> Russell Bryant >>> Digium, Inc. | Engineering Manager, Open Source Software >>> 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA >>> www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org >> >> >> This could be an issue with the CA keys used in Safari. I remember >> having to chain load a root key for a server just for iphone support a >> while back. looking >> >> Apache option is "SSLCertificateChainFile /full/path/to/your.ca-bundle" > > Can Safari open a connection to https://issues.asterisk.org? (no /jira suffix) > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Question on how many phones
Can a quad or six core server with 4 GIG RAM running asterisk 1.4 handle 1000 polycom phones. 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] How asterisk use pri channel
> We have two pri line and I want to see how asterisk distribute > outgoing call per channels > > I meant it use first last channel 47 or it will use first channel? > > Or it will allocate dynamically ? Extracted from chan_dahdi.c: Dial(DAHDI/pseudo[/extension[/options]]) Dial(DAHDI/[c|r|d][/extension[/options]]) Dial(DAHDI/![c|r|d][/extension[/options]]) Dial(DAHDI/i[/extension[/options]]) Dial(DAHDI/[i-](g|G|r|R)[c|r|d][/extension[/options]]) i - ISDN span channel restriction. Used by CC to ensure that the CC recall goes out the same span. Also to make ISDN channel names dialable when the sequence number is stripped off. (Used by DTMF attended transfer feature.) g - channel group allocation search forward G - channel group allocation search backward r - channel group allocation round robin search forward R - channel group allocation round robin search backward c - Wait for DTMF digit to confirm answer r - Set distintive ring cadance number d - Force bearer capability for ISDN/SS7 call to digital. All are valid for v1.8 and trunk. The i option and ! option are not valid earlier than v1.8. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How asterisk use pri channel
Hi, We have two pri line and I want to see how asterisk distribute outgoing call per channels I meant it use first last channel 47 or it will use first channel? Or it will allocate dynamically ? -- Sent from my iPhone -- _ -- 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] issues.asterisk.org/jira not working
On Wed, Jun 8, 2011 at 5:56 PM, Satish Patel wrote: > > It not working on iPhone. It's saying not able to make secure connection > > -- > Sent from my iPhone Satish, Can you share what the SSL/TLS Cert says? Safari and mobile platforms have a smaller list of CAs, just to make life hard for us sysadmin types... -- ~~~ Andrew "lathama" Latham lath...@gmail.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] issues.asterisk.org/jira not working
It not working on iPhone. It's saying not able to make secure connection -- Sent from my iPhone On Jun 8, 2011, at 4:54 PM, "Kevin P. Fleming" wrote: On 06/08/2011 02:27 PM, Andrew Latham wrote: On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant wrote: A number of people are reporting that Safari is not working properly with JIRA. Use Firefox or Chrome for now. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org This could be an issue with the CA keys used in Safari. I remember having to chain load a root key for a server just for iphone support a while back. looking Apache option is "SSLCertificateChainFile /full/path/to/your.ca- bundle" Can Safari open a connection to https://issues.asterisk.org? (no / jira suffix) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] LXC and Dahdi
Howdy, I am playing around with asterisk within an LXC container on Ubuntu 11.04. I have asterisk (1.4.42) running fine, but want access to dahdi_dummy for timing (meetme). I have dahdi installed on the "host", and dahdi_dummy is loaded: root@astnorth:/# ls -ltr /dev/dahdi total 0 crw-rw 1 root root 196, 250 2011-06-08 13:59 transcode crw-rw 1 root root 196, 253 2011-06-08 13:59 timer crw-rw 1 root root 196, 255 2011-06-08 13:59 pseudo crw-rw 1 root root 196, 0 2011-06-08 13:59 ctl crw-rw 1 root root 196, 254 2011-06-08 13:59 channel root@astnorth:/# But in the container I don't see them: root@artha:/# ls -ltr /dev/dahdi total 0 even though /dev/dahdi showed up as soon as I loaded the kernel module in the host. Maybe this is more a question for an LXC list, but I noted that a few people had played with this (asterisk in LXC) and wondered if anyone managed to get dahdi_dummy shared across their containers... Thanks for any input, -- Jeff LaCoursiere SunFone 340-715-7600 x222 j...@sunfone.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] issues.asterisk.org/jira not working
I'm using firefox and now it's works befrore after fill out information submit I got blank page. -- Sent from my iPhone On Jun 8, 2011, at 4:54 PM, "Kevin P. Fleming" wrote: On 06/08/2011 02:27 PM, Andrew Latham wrote: On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant wrote: A number of people are reporting that Safari is not working properly with JIRA. Use Firefox or Chrome for now. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org This could be an issue with the CA keys used in Safari. I remember having to chain load a root key for a server just for iphone support a while back. looking Apache option is "SSLCertificateChainFile /full/path/to/your.ca- bundle" Can Safari open a connection to https://issues.asterisk.org? (no / jira suffix) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] issues.asterisk.org/jira not working
On 06/08/2011 02:27 PM, Andrew Latham wrote: On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant wrote: A number of people are reporting that Safari is not working properly with JIRA. Use Firefox or Chrome for now. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org This could be an issue with the CA keys used in Safari. I remember having to chain load a root key for a server just for iphone support a while back. looking Apache option is "SSLCertificateChainFile /full/path/to/your.ca-bundle" Can Safari open a connection to https://issues.asterisk.org? (no /jira suffix) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] After wiki.asterisk.org was upgraded my user no loger exists.
On 11-06-08 10:34 AM, Paul Belanger wrote: On 11-06-07 10:20 PM, Jose P. Espinal wrote: Hello Guys, After the Wiki was updated to the 3.5.X version, my username is no loger available: user: khratos mail: j...@slackware-es.com I had some documents on my personal space. Is there a way to recover the account? Yes, My account is also missing, I believe there is an issue with crowd and something is scheduled to look at it. Do you mind checking again? I'm now able to access my account again. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://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] issues.asterisk.org/jira not working
On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant wrote: > A number of people are reporting that Safari is not working properly with > JIRA. Use Firefox or Chrome for now. > > -- > Russell Bryant > Digium, Inc. | Engineering Manager, Open Source Software > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org This could be an issue with the CA keys used in Safari. I remember having to chain load a root key for a server just for iphone support a while back. looking Apache option is "SSLCertificateChainFile /full/path/to/your.ca-bundle" -- ~~~ Andrew "lathama" Latham lath...@gmail.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] issues.asterisk.org/jira not working
A number of people are reporting that Safari is not working properly with JIRA. Use Firefox or Chrome for now. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org - Original Message - > I get this on my Mac: > > > > > Safari can’t open the page. > > Safari can’t open the page > “https://issues.asterisk.org/jira/browse/ASTERISK-17984” because > Safari can’t establish a secure connection to the server > “issues.asterisk.org”. > > > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > > CfMC > http://www.cfmc.com/ > > > > > > On Jun 8, 2011, at 11:38 AM, William Stillwell wrote: > > > > > > You mean this one? > > https://issues.asterisk.org/jira/browse/ASTERISK-17984 > > > > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish > patel > Sent: Wednesday, June 08, 2011 2:17 PM > To: asterisk-users > Subject: [asterisk-users] issues.asterisk.org/jira not working > > > Bad day today. Why this new JIRA system not working. I have created > issue and submit and i got blank page.. Please someone help me to > create BUG!!! -- > _ > -- 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] issues.asterisk.org/jira not working
I get this on my Mac: Safari can’t open the page. Safari can’t open the page “https://issues.asterisk.org/jira/browse/ASTERISK-17984” because Safari can’t establish a secure connection to the server “issues.asterisk.org”. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 8, 2011, at 11:38 AM, William Stillwell wrote: > You mean this one? > > https://issues.asterisk.org/jira/browse/ASTERISK-17984 > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel > Sent: Wednesday, June 08, 2011 2:17 PM > To: asterisk-users > Subject: [asterisk-users] issues.asterisk.org/jira not working > > Bad day today. Why this new JIRA system not working. I have created issue > and submit and i got blank page.. Please someone help me to create > BUG!!! > > -- > _ > -- 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] No IVR listen at device end......SIP phone is working fine
On Thu, 9 Jun 2011, virendra bhati wrote: When we make calls into asterisk with the help of our mobile, landline number, Cisco 79XX series then we didn't able to here any IVR which is playing into asterisk server. But when we dial from SIP softphone then all is working fine and we are able to here the IVR sound files. What is the problem in this case please help me.. NAT is a frequent culprit. Firewalls and iptables are also suspect. Better details, better answers. -- 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] how asterisk work with VoIP trunk?
On Wed, 8 Jun 2011, virendra bhati wrote: I have working experience of asterisk with PRI lines. Recently I have took VoIP routes from my provider. My basic issue is that now how asterisk will behave in such case. I mean in PRI call will come as below process PRI - -> Digium Card - -> Dadhi/Zap - -> Extensions.conf What will be the VoIP calling call flow in Incoming and outgoing calls? Eth[x] -> sip.conf -> extensions.conf -- 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
[asterisk-users] No IVR listen at device end......SIP phone is working fine
Hi List, When we make calls into asterisk with the help of our mobile, landline number, Cisco 79XX series then we didn't able to here any IVR which is playing into asterisk server. But when we dial from SIP softphone then all is working fine and we are able to here the IVR sound files. What is the problem in this case please help me.. -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] issues.asterisk.org/jira not working
You mean this one? https://issues.asterisk.org/jira/browse/ASTERISK-17984 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 2:17 PM To: asterisk-users Subject: [asterisk-users] issues.asterisk.org/jira not working Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!! -- _ -- 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] CallerID issue
Hi List, I am making outgoing call from asterisk to GSM network with the help of VoIP trunk(SIP trunk) then I am not geting any caller ID at destination end. Is this the asterisk issue or VoIP trunk issue? Is this is due to asterisk then how we solve it? I already user Set(CALLERID(num)=XXX) in dialplan. - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how asterisk work with VoIP trunk?
Hi List, I have working experience of asterisk with PRI lines. Recently I have took VoIP routes from my provider. My basic issue is that now how asterisk will behave in such case. I mean in PRI call will come as below process PRI - -> Digium Card - -> Dadhi/Zap - -> Extensions.conf What will be the VoIP calling call flow in Incoming and outgoing calls? -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk User -- _ -- 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] issues.asterisk.org/jira not working
Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!! -- _ -- 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 and Audiocodes PRI card
This card is a standalone SIP media server on a PCI blade. But you can make it work with Asterisk for that you have to tweak Asterisk source and as well as you have to buy API from audiocodes if I am not wrong. Instead of this why can't you use Sangoma or Digium cards? On Wed, Jun 8, 2011 at 11:39 PM, Jonas Kellens wrote: > Hello list, > > can anyone tell me if this card : > > http://www.audiocodes.com/product/ipm-260-sip > > is compatible with Asterisk (DAHDI) for use as PCI PRI card ? > > > > Kind regards, > Jonas. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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 and Audiocodes PRI card
Hello list, can anyone tell me if this card : http://www.audiocodes.com/product/ipm-260-sip is compatible with Asterisk (DAHDI) for use as PCI PRI card ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interesting PRI issue
Hey Guys!
Please help me to find out issue. I have two PRI
## Span 1: WPT1/0 "wanpipe1 card 0"
span=1,1,0,esf,b8zs
bchan=1-23
hardhdlc=24
echocanceller=mg2,1-23
## Span 2: WPT1/1 "wanpipe2 card 1"
span=2,2,0,esf,b8zs
bchan=25-47
hardhdlc=48
echocanceller=mg2,25-47
Sometime my calls got through but some time i am getting pri cause 44
sebpbx1*CLI>
== Using SIP RTP CoS mark 5
-- Executing [6463279153@from-sip:1] Dial("SIP/8227-02b1",
"DAHDI/G1/16463279153") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called DAHDI/G1/16463279153
-- Span 2: Channel 0/23 got hangup, cause 44
-- Span 2: Forcing restart of channel 0/23 since channel reported in use
-- Hungup 'DAHDI/i2/16463279153-fe'
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/8227-02b1' status is 'CHANUNAVAIL'
-- Span 2: Channel 0/23 successfully restarted
--
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Re: [asterisk-users] Asterisk 1.8 broken MWI
On 8 June 2011 17:20, satish patel wrote: > Interesting thing is when i reload sip.conf i got MWI lamp working on > polycom 501 > > But its not working when anyone leave voicemail. Do you know its some > timeout or polling setting in sip.conf ? > > Still my question is my my asterisk not sending NOTIFY message ? Do i need > to subscribe my phone to asterisk ? > Does this help? https://issues.asterisk.org/jira/browse/ASTERISK-17866 Regards, Steve -- _ -- 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] R: Re: Looking for Email to Fax Solutions
What do you think to do with the solution ? Cause we developed it ourselves and is in run on more company, if you want I can talk you about it. In any case, Avantfax I remember to be a frontend for Hilafax. I don't know the other one, sorry Enrico Cicconi www.rdmnet.it Cordialmente Enrico Cicconi -Original Message- From: "Paddy Grice" Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 8 Jun 2011 17:23:41 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Reply-To: pa...@wizaner.com, Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Looking for Email to Fax Solutions On 06/08/2011 01:09 AM, Paddy Grice wrote: > Hi All > > I am looking for a small scale Email to fax solution > > Searches seem to throw up > > AsterFax http://sourceforge.net/projects/asterfax/ which seems to go > to http://www.noojee.com.au/products/noojee-fax/fax-overview/ > email12fax http://wpkg.org/email2fax/index.php/Main_Page > > I would appreciate any comments on these or other solutions > > I am running asterisk 1.4 and I am looking for a small scale solution > say 10 lines (ddis) While I designed it with Asterisk 1.6 or 1.8 in mind, you may try this: http://messinet.com/trac/wiki/AsteriskFAXGateway I have some time next week if it needs some tweaks to work with Asterisk 1.4. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Looks like it will certainly help me - I will work through it and let you know - I am away for a few days so will be next week before I can try it out. Thanks Paddy -- _ -- 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] Looking for Email to Fax Solutions
On 06/08/2011 01:09 AM, Paddy Grice wrote: > Hi All > > I am looking for a small scale Email to fax solution > > Searches seem to throw up > > AsterFax http://sourceforge.net/projects/asterfax/ which seems to go > to http://www.noojee.com.au/products/noojee-fax/fax-overview/ > email12fax http://wpkg.org/email2fax/index.php/Main_Page > > I would appreciate any comments on these or other solutions > > I am running asterisk 1.4 and I am looking for a small scale solution > say 10 lines (ddis) While I designed it with Asterisk 1.6 or 1.8 in mind, you may try this: http://messinet.com/trac/wiki/AsteriskFAXGateway I have some time next week if it needs some tweaks to work with Asterisk 1.4. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Looks like it will certainly help me - I will work through it and let you know - I am away for a few days so will be next week before I can try it out. Thanks Paddy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 broken MWI
Interesting thing is when i reload sip.conf i got MWI lamp working on polycom 501 But its not working when anyone leave voicemail. Do you know its some timeout or polling setting in sip.conf ? Still my question is my my asterisk not sending NOTIFY message ? Do i need to subscribe my phone to asterisk ? From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 15:38:53 + Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI >> Yes its under [defailt] section at voicemail.conf Sorry it my typo error. >>When there is a new message in a mailbox, does "voicemail show users" show >>new messages for that mailbox? Yes, I can see there are 10 voicemail root@campbx1:~# asterisk -rx 'voicemail show users' | grep -i 7623 default7623 Satish Patel 10 > From: ewiel...@nyigc.com > To: asterisk-users@lists.digium.com > Date: Wed, 8 Jun 2011 11:33:31 -0400 > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > I assume you misspelled "default" in your e-mail and not voicemail.conf. If > not, that is your problem. > > When there is a new message in a mailbox, does "voicemail show users" show > new messages for that mailbox? > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > satish patel > > Sent: Wednesday, June 08, 2011 11:21 AM > > To: asterisk-users > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > Yes its under [defailt] section at voicemail.conf > > > > > From: ewiel...@nyigc.com > > > To: asterisk-users@lists.digium.com > > > Date: Wed, 8 Jun 2011 11:17:26 -0400 > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > Is 7623 listed in voicemail.conf under the [default] section? > > > > > > > -Original Message- > > > > From: asterisk-users-boun...@lists.digium.com > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > satish patel > > > > Sent: Wednesday, June 08, 2011 11:15 AM > > > > To: asterisk-users > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > I do have that > > > > > > > > sip.conf > > > > > > > > [7623](cam-exten) > > > > callerid="Satish Patel" <7623> > > > > accountcode="Satish Patel" > > > > mailbox=7623@default > > > > > > > > > > > > > From: ewiel...@nyigc.com > > > > > To: asterisk-users@lists.digium.com > > > > > Date: Wed, 8 Jun 2011 11:03:24 -0400 > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 > > > > tarball. Make sure your mailboxes specify a voicemail context > > > > on each mailbox= line. > > > > > > > > > > > -Original Message- > > > > > > From: asterisk-users-boun...@lists.digium.com > > > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > > > satish patel > > > > > > Sent: Wednesday, June 08, 2011 10:44 AM > > > > > > To: asterisk-users > > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > Truly speaking, I went though that file and i found nothing > > > > > > in that file related major changes. It was working perfect > > > > > > before 1.2 > > > > > > > > > > > > May be i am missing some configuration option. Do you know > > > > > > any debug method to make it work ? > > > > > > > > > > > > > From: ewiel...@nyigc.com > > > > > > > To: asterisk-users@lists.digium.com > > > > > > > Date: Wed, 8 Jun 2011 10:34:16 -0400 > > > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > > > All major changes are listed in the UPGRADE.txt files > > > > > > included in the 1.8 tarball. > > > > > > > > > > > > > > > -Original Message- > > > > > > > > From: asterisk-users-boun...@lists.digium.com > > > > > > > > [mailto:asterisk-users-boun...@lists.digium.com] > > On Behalf Of > > > > > > > > satish patel > > > > > > > > Sent: Wednesday, June 08, 2011 9:57 AM > > > > > > > > To: asterisk-users > > > > > > > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > > > > > Hi ALL, > > > > > > > > > > > > > > > > After upgrade 1.8 my MWI wasn't working I do have > > setting in > > > > > > > > voicemail.conf. Do i need to do anything else to > > fix my MWI > > > > > > > > on polycom 501 ? It was working with 1.2 asterisk. > > > > > > > > > > > > > > > > pollmailboxes=yes > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > > > > > > > > > > > > > > > _ > > > > > > > -- 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: > > > > >
[asterisk-users] Update problem | CLI commands missing
Hi List, I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW. Currently I'm running Asterisk 1.8.3.3. I have the following problem, if I do the update to the actual 1.8.4.2. There are several commands on the CLI which are not working or even not present like core show uptime (not working) core restart (not present) core show version (not present) my Skype for Asterisk is also not loaded correctly. 190 modules are loaded, if I do a 'module show'. I miss also some messages in the log like "[Jun 7 21:21:31] VERBOSE[3449] loader.c: func_version.so => (Get Asterisk Version/Build Info)". Does anyone know something about this problem? best regards Christoph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 broken MWI
>> Yes its under [defailt] section at voicemail.conf Sorry it my typo error. >>When there is a new message in a mailbox, does "voicemail show users" show >>new messages for that mailbox? Yes, I can see there are 10 voicemail root@campbx1:~# asterisk -rx 'voicemail show users' | grep -i 7623 default7623 Satish Patel 10 > From: ewiel...@nyigc.com > To: asterisk-users@lists.digium.com > Date: Wed, 8 Jun 2011 11:33:31 -0400 > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > I assume you misspelled "default" in your e-mail and not voicemail.conf. If > not, that is your problem. > > When there is a new message in a mailbox, does "voicemail show users" show > new messages for that mailbox? > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > satish patel > > Sent: Wednesday, June 08, 2011 11:21 AM > > To: asterisk-users > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > Yes its under [defailt] section at voicemail.conf > > > > > From: ewiel...@nyigc.com > > > To: asterisk-users@lists.digium.com > > > Date: Wed, 8 Jun 2011 11:17:26 -0400 > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > Is 7623 listed in voicemail.conf under the [default] section? > > > > > > > -Original Message- > > > > From: asterisk-users-boun...@lists.digium.com > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > satish patel > > > > Sent: Wednesday, June 08, 2011 11:15 AM > > > > To: asterisk-users > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > I do have that > > > > > > > > sip.conf > > > > > > > > [7623](cam-exten) > > > > callerid="Satish Patel" <7623> > > > > accountcode="Satish Patel" > > > > mailbox=7623@default > > > > > > > > > > > > > From: ewiel...@nyigc.com > > > > > To: asterisk-users@lists.digium.com > > > > > Date: Wed, 8 Jun 2011 11:03:24 -0400 > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 > > > > tarball. Make sure your mailboxes specify a voicemail context > > > > on each mailbox= line. > > > > > > > > > > > -Original Message- > > > > > > From: asterisk-users-boun...@lists.digium.com > > > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > > > satish patel > > > > > > Sent: Wednesday, June 08, 2011 10:44 AM > > > > > > To: asterisk-users > > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > Truly speaking, I went though that file and i found nothing > > > > > > in that file related major changes. It was working perfect > > > > > > before 1.2 > > > > > > > > > > > > May be i am missing some configuration option. Do you know > > > > > > any debug method to make it work ? > > > > > > > > > > > > > From: ewiel...@nyigc.com > > > > > > > To: asterisk-users@lists.digium.com > > > > > > > Date: Wed, 8 Jun 2011 10:34:16 -0400 > > > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > > > All major changes are listed in the UPGRADE.txt files > > > > > > included in the 1.8 tarball. > > > > > > > > > > > > > > > -Original Message- > > > > > > > > From: asterisk-users-boun...@lists.digium.com > > > > > > > > [mailto:asterisk-users-boun...@lists.digium.com] > > On Behalf Of > > > > > > > > satish patel > > > > > > > > Sent: Wednesday, June 08, 2011 9:57 AM > > > > > > > > To: asterisk-users > > > > > > > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > > > > > Hi ALL, > > > > > > > > > > > > > > > > After upgrade 1.8 my MWI wasn't working I do have > > setting in > > > > > > > > voicemail.conf. Do i need to do anything else to > > fix my MWI > > > > > > > > on polycom 501 ? It was working with 1.2 asterisk. > > > > > > > > > > > > > > > > pollmailboxes=yes > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > > > > > > > > > > > > > > > _ > > > > > > > -- 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 U
Re: [asterisk-users] Asterisk 1.8 broken MWI
I assume you misspelled "default" in your e-mail and not voicemail.conf. If not, that is your problem. When there is a new message in a mailbox, does "voicemail show users" show new messages for that mailbox? > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > satish patel > Sent: Wednesday, June 08, 2011 11:21 AM > To: asterisk-users > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > Yes its under [defailt] section at voicemail.conf > > > From: ewiel...@nyigc.com > > To: asterisk-users@lists.digium.com > > Date: Wed, 8 Jun 2011 11:17:26 -0400 > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > Is 7623 listed in voicemail.conf under the [default] section? > > > > > -Original Message- > > > From: asterisk-users-boun...@lists.digium.com > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > satish patel > > > Sent: Wednesday, June 08, 2011 11:15 AM > > > To: asterisk-users > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > I do have that > > > > > > sip.conf > > > > > > [7623](cam-exten) > > > callerid="Satish Patel" <7623> > > > accountcode="Satish Patel" > > > mailbox=7623@default > > > > > > > > > > From: ewiel...@nyigc.com > > > > To: asterisk-users@lists.digium.com > > > > Date: Wed, 8 Jun 2011 11:03:24 -0400 > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 > > > tarball. Make sure your mailboxes specify a voicemail context > > > on each mailbox= line. > > > > > > > > > -Original Message- > > > > > From: asterisk-users-boun...@lists.digium.com > > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > > satish patel > > > > > Sent: Wednesday, June 08, 2011 10:44 AM > > > > > To: asterisk-users > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > Truly speaking, I went though that file and i found nothing > > > > > in that file related major changes. It was working perfect > > > > > before 1.2 > > > > > > > > > > May be i am missing some configuration option. Do you know > > > > > any debug method to make it work ? > > > > > > > > > > > From: ewiel...@nyigc.com > > > > > > To: asterisk-users@lists.digium.com > > > > > > Date: Wed, 8 Jun 2011 10:34:16 -0400 > > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > All major changes are listed in the UPGRADE.txt files > > > > > included in the 1.8 tarball. > > > > > > > > > > > > > -Original Message- > > > > > > > From: asterisk-users-boun...@lists.digium.com > > > > > > > [mailto:asterisk-users-boun...@lists.digium.com] > On Behalf Of > > > > > > > satish patel > > > > > > > Sent: Wednesday, June 08, 2011 9:57 AM > > > > > > > To: asterisk-users > > > > > > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > > > Hi ALL, > > > > > > > > > > > > > > After upgrade 1.8 my MWI wasn't working I do have > setting in > > > > > > > voicemail.conf. Do i need to do anything else to > fix my MWI > > > > > > > on polycom 501 ? It was working with 1.2 asterisk. > > > > > > > > > > > > > > pollmailboxes=yes > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > > > > > > > > > > > _ > > > > > > -- 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 > > -- _ -- 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.or
[asterisk-users] call transfer back to a sourcing switch
If call comes into PBX-A and based on the DNIS it comes into my box PBX-B my box then says ring phone C. Person answers. They want to transfer the call to a phone going back out PBX-A. All this is fine of course. my question is when phone C transfers the call is there a way PBX-B can drop out of the mix. 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] Asterisk 1.8 broken MWI
Yes its under [defailt] section at voicemail.conf > From: ewiel...@nyigc.com > To: asterisk-users@lists.digium.com > Date: Wed, 8 Jun 2011 11:17:26 -0400 > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > Is 7623 listed in voicemail.conf under the [default] section? > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > satish patel > > Sent: Wednesday, June 08, 2011 11:15 AM > > To: asterisk-users > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > I do have that > > > > sip.conf > > > > [7623](cam-exten) > > callerid="Satish Patel" <7623> > > accountcode="Satish Patel" > > mailbox=7623@default > > > > > > > From: ewiel...@nyigc.com > > > To: asterisk-users@lists.digium.com > > > Date: Wed, 8 Jun 2011 11:03:24 -0400 > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 > > tarball. Make sure your mailboxes specify a voicemail context > > on each mailbox= line. > > > > > > > -Original Message- > > > > From: asterisk-users-boun...@lists.digium.com > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > satish patel > > > > Sent: Wednesday, June 08, 2011 10:44 AM > > > > To: asterisk-users > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > Truly speaking, I went though that file and i found nothing > > > > in that file related major changes. It was working perfect > > > > before 1.2 > > > > > > > > May be i am missing some configuration option. Do you know > > > > any debug method to make it work ? > > > > > > > > > From: ewiel...@nyigc.com > > > > > To: asterisk-users@lists.digium.com > > > > > Date: Wed, 8 Jun 2011 10:34:16 -0400 > > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > All major changes are listed in the UPGRADE.txt files > > > > included in the 1.8 tarball. > > > > > > > > > > > -Original Message- > > > > > > From: asterisk-users-boun...@lists.digium.com > > > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > > > satish patel > > > > > > Sent: Wednesday, June 08, 2011 9:57 AM > > > > > > To: asterisk-users > > > > > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > > > Hi ALL, > > > > > > > > > > > > After upgrade 1.8 my MWI wasn't working I do have setting in > > > > > > voicemail.conf. Do i need to do anything else to fix my MWI > > > > > > on polycom 501 ? It was working with 1.2 asterisk. > > > > > > > > > > > > pollmailboxes=yes > > > > > > > > > > > > > > > > > > > > > > -- > > > > > > > > > > > _ > > > > > -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 broken MWI
Do you think i should enable ? ; searchcontexts=yes > From: ewiel...@nyigc.com > To: asterisk-users@lists.digium.com > Date: Wed, 8 Jun 2011 11:03:24 -0400 > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make > sure your mailboxes specify a voicemail context on each mailbox= line. > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > satish patel > > Sent: Wednesday, June 08, 2011 10:44 AM > > To: asterisk-users > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > Truly speaking, I went though that file and i found nothing > > in that file related major changes. It was working perfect > > before 1.2 > > > > May be i am missing some configuration option. Do you know > > any debug method to make it work ? > > > > > From: ewiel...@nyigc.com > > > To: asterisk-users@lists.digium.com > > > Date: Wed, 8 Jun 2011 10:34:16 -0400 > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > All major changes are listed in the UPGRADE.txt files > > included in the 1.8 tarball. > > > > > > > -Original Message- > > > > From: asterisk-users-boun...@lists.digium.com > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > satish patel > > > > Sent: Wednesday, June 08, 2011 9:57 AM > > > > To: asterisk-users > > > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > Hi ALL, > > > > > > > > After upgrade 1.8 my MWI wasn't working I do have setting in > > > > voicemail.conf. Do i need to do anything else to fix my MWI > > > > on polycom 501 ? It was working with 1.2 asterisk. > > > > > > > > pollmailboxes=yes > > > > > > > > > > > > > > -- > > > > > _ > > > -- 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] Asterisk 1.8 broken MWI
Is 7623 listed in voicemail.conf under the [default] section? > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > satish patel > Sent: Wednesday, June 08, 2011 11:15 AM > To: asterisk-users > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > I do have that > > sip.conf > > [7623](cam-exten) > callerid="Satish Patel" <7623> > accountcode="Satish Patel" > mailbox=7623@default > > > > From: ewiel...@nyigc.com > > To: asterisk-users@lists.digium.com > > Date: Wed, 8 Jun 2011 11:03:24 -0400 > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 > tarball. Make sure your mailboxes specify a voicemail context > on each mailbox= line. > > > > > -Original Message- > > > From: asterisk-users-boun...@lists.digium.com > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > satish patel > > > Sent: Wednesday, June 08, 2011 10:44 AM > > > To: asterisk-users > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > Truly speaking, I went though that file and i found nothing > > > in that file related major changes. It was working perfect > > > before 1.2 > > > > > > May be i am missing some configuration option. Do you know > > > any debug method to make it work ? > > > > > > > From: ewiel...@nyigc.com > > > > To: asterisk-users@lists.digium.com > > > > Date: Wed, 8 Jun 2011 10:34:16 -0400 > > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > All major changes are listed in the UPGRADE.txt files > > > included in the 1.8 tarball. > > > > > > > > > -Original Message- > > > > > From: asterisk-users-boun...@lists.digium.com > > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > > satish patel > > > > > Sent: Wednesday, June 08, 2011 9:57 AM > > > > > To: asterisk-users > > > > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > > > Hi ALL, > > > > > > > > > > After upgrade 1.8 my MWI wasn't working I do have setting in > > > > > voicemail.conf. Do i need to do anything else to fix my MWI > > > > > on polycom 501 ? It was working with 1.2 asterisk. > > > > > > > > > > pollmailboxes=yes > > > > > > > > > > > > > > > > > > -- > > > > > > > > _ > > > > -- 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] Asterisk 1.8 broken MWI
I do have that sip.conf [7623](cam-exten) callerid="Satish Patel" <7623> accountcode="Satish Patel" mailbox=7623@default > From: ewiel...@nyigc.com > To: asterisk-users@lists.digium.com > Date: Wed, 8 Jun 2011 11:03:24 -0400 > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make > sure your mailboxes specify a voicemail context on each mailbox= line. > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > satish patel > > Sent: Wednesday, June 08, 2011 10:44 AM > > To: asterisk-users > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > Truly speaking, I went though that file and i found nothing > > in that file related major changes. It was working perfect > > before 1.2 > > > > May be i am missing some configuration option. Do you know > > any debug method to make it work ? > > > > > From: ewiel...@nyigc.com > > > To: asterisk-users@lists.digium.com > > > Date: Wed, 8 Jun 2011 10:34:16 -0400 > > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > All major changes are listed in the UPGRADE.txt files > > included in the 1.8 tarball. > > > > > > > -Original Message- > > > > From: asterisk-users-boun...@lists.digium.com > > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > > satish patel > > > > Sent: Wednesday, June 08, 2011 9:57 AM > > > > To: asterisk-users > > > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > > > Hi ALL, > > > > > > > > After upgrade 1.8 my MWI wasn't working I do have setting in > > > > voicemail.conf. Do i need to do anything else to fix my MWI > > > > on polycom 501 ? It was working with 1.2 asterisk. > > > > > > > > pollmailboxes=yes > > > > > > > > > > > > > > -- > > > > > _ > > > -- 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] Asterisk 1.8 broken MWI
Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make sure your mailboxes specify a voicemail context on each mailbox= line. > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > satish patel > Sent: Wednesday, June 08, 2011 10:44 AM > To: asterisk-users > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > Truly speaking, I went though that file and i found nothing > in that file related major changes. It was working perfect > before 1.2 > > May be i am missing some configuration option. Do you know > any debug method to make it work ? > > > From: ewiel...@nyigc.com > > To: asterisk-users@lists.digium.com > > Date: Wed, 8 Jun 2011 10:34:16 -0400 > > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > > > All major changes are listed in the UPGRADE.txt files > included in the 1.8 tarball. > > > > > -Original Message- > > > From: asterisk-users-boun...@lists.digium.com > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > > satish patel > > > Sent: Wednesday, June 08, 2011 9:57 AM > > > To: asterisk-users > > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > > > Hi ALL, > > > > > > After upgrade 1.8 my MWI wasn't working I do have setting in > > > voicemail.conf. Do i need to do anything else to fix my MWI > > > on polycom 501 ? It was working with 1.2 asterisk. > > > > > > pollmailboxes=yes > > > > > > > > > > -- > > > _ > > -- 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 1.8 broken MWI
Following is my debug and look like its not sending MWI NOTIFY message to phone Reliably Transmitting (no NAT) to 172.30.245.143:5060: OPTIONS sip:7623@172.30.245.143 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3 Max-Forwards: 70 From: "asterisk" ;tag=as26352734 To: Contact: Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r321926 Date: Wed, 08 Jun 2011 14:49:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.30.245.143:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3 From: "asterisk" ;tag=as26352734 To: ;tag=E777D3B9-F605D562 CSeq: 102 OPTIONS Call-ID: 44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Content-Length: 0 <-> --- (10 headers 0 lines) --- Really destroying SIP dialog '44c3ac7b4c37786c3fda41e12d1a907c@172.30.1.46:5060' Method: OPTIONS Reliably Transmitting (no NAT) to 172.30.245.143:5060: OPTIONS sip:7623@172.30.245.143 SIP/2.0 Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK18a12f37 Max-Forwards: 70 From: "asterisk" ;tag=as0c8778f4 To: Contact: Call-ID: 50d5cc4d5510ae014c6641702faea18d@172.30.1.46:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-branch-1.8-r321926 Date: Wed, 08 Jun 2011 14:50:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:172.30.245.143:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK18a12f37 From: "asterisk" ;tag=as0c8778f4 To: ;tag=47557FCE-869CEA2F CSeq: 102 OPTIONS Call-ID: 50d5cc4d5510ae014c6641702faea18d@172.30.1.46:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043 Content-Length: 0 <-> --- (10 headers 0 lines) --- From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 14:43:57 + Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? > From: ewiel...@nyigc.com > To: asterisk-users@lists.digium.com > Date: Wed, 8 Jun 2011 10:34:16 -0400 > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > All major changes are listed in the UPGRADE.txt files included in the 1.8 > tarball. > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > satish patel > > Sent: Wednesday, June 08, 2011 9:57 AM > > To: asterisk-users > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > Hi ALL, > > > > After upgrade 1.8 my MWI wasn't working I do have setting in > > voicemail.conf. Do i need to do anything else to fix my MWI > > on polycom 501 ? It was working with 1.2 asterisk. > > > > pollmailboxes=yes > > > > > > -- > _ > -- 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] Looking for Email to Fax Solutions
On 06/08/2011 01:09 AM, Paddy Grice wrote: > Hi All > > I am looking for a small scale Email to fax solution > > Searches seem to throw up > > AsterFax http://sourceforge.net/projects/asterfax/ which seems to go to > http://www.noojee.com.au/products/noojee-fax/fax-overview/ > email12fax http://wpkg.org/email2fax/index.php/Main_Page > > I would appreciate any comments on these or other solutions > > I am running asterisk 1.4 and I am looking for a small scale solution say 10 > lines (ddis) While I designed it with Asterisk 1.6 or 1.8 in mind, you may try this: http://messinet.com/trac/wiki/AsteriskFAXGateway I have some time next week if it needs some tweaks to work with Asterisk 1.4. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 broken MWI
Truly speaking, I went though that file and i found nothing in that file related major changes. It was working perfect before 1.2 May be i am missing some configuration option. Do you know any debug method to make it work ? > From: ewiel...@nyigc.com > To: asterisk-users@lists.digium.com > Date: Wed, 8 Jun 2011 10:34:16 -0400 > Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI > > All major changes are listed in the UPGRADE.txt files included in the 1.8 > tarball. > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > > satish patel > > Sent: Wednesday, June 08, 2011 9:57 AM > > To: asterisk-users > > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > > > Hi ALL, > > > > After upgrade 1.8 my MWI wasn't working I do have setting in > > voicemail.conf. Do i need to do anything else to fix my MWI > > on polycom 501 ? It was working with 1.2 asterisk. > > > > pollmailboxes=yes > > > > > > -- > _ > -- 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 1.8 broken MWI
All major changes are listed in the UPGRADE.txt files included in the 1.8 tarball. > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > satish patel > Sent: Wednesday, June 08, 2011 9:57 AM > To: asterisk-users > Subject: [asterisk-users] Asterisk 1.8 broken MWI > > Hi ALL, > > After upgrade 1.8 my MWI wasn't working I do have setting in > voicemail.conf. Do i need to do anything else to fix my MWI > on polycom 501 ? It was working with 1.2 asterisk. > > pollmailboxes=yes > > -- _ -- 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] After wiki.asterisk.org was upgraded my user no loger exists.
On 11-06-07 10:20 PM, Jose P. Espinal wrote: Hello Guys, After the Wiki was updated to the 3.5.X version, my username is no loger available: user: khratos mail: j...@slackware-es.com I had some documents on my personal space. Is there a way to recover the account? Yes, My account is also missing, I believe there is an issue with crowd and something is scheduled to look at it. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://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
[asterisk-users] Asterisk 1.8 broken MWI
Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -- _ -- 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 hangup request, cause 18
We have two sites. BOSTON and California We are having only issue with California PRI line related cause 18 but BOSTON pri has no issue. All settings are same on both Asterisk. Today i will talk to service provider and will see. pridialplan=uknown fixed many issues except cause 18 -S > Date: Wed, 8 Jun 2011 15:41:04 +0200 > From: t...@ovm-group.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] PRI hangup request, cause 18 > > Ist the same operator connected to the pri-line? Perhaps another > telco-operator can not connect to the desired destination - for whatever > reason. > > Am 08.06.2011 12:55, schrieb Satish Patel: > > Thanks for reply, > > > > But I'm able to call those number from my cell phone and othere pri. > > > > I'm only having this issue on 2 pri line rest are working ? > > > > -- > > Sent from my iPhone > > > > On Jun 8, 2011, at 5:44 AM, Doug Lytle wrote: > > > >> satish patel wrote: > >>> We are getting hangup cause 18 > >> > >> http://networking.ringofsaturn.com/Routers/isdncausecodes.php > >> > >> *Cause No. 18 - no user responding.* > >> This cause is used when a called party does not respond to a call > >> establishment message with either an alerting or connect indication > >> within the prescribed period of time allocated. > >> > >> What it means: > >> The equipment on the other end does not answer the call. Usually this > >> is a misconfiguration on the equipment being called. > >> > >> > >> > >> Doug > >> > >> -- > >> Ben Franklin quote: > >> > >> "Those who would give up Essential Liberty to purchase a little > >> Temporary Safety, deserve neither Liberty nor Safety." > >> > >> > >> -- > >> _ > >> -- 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 > > -- > Thorsten Göllner > > OVM Office Voice Media GmbH > Herderstrasse 68 > 40237 Düsseldorf > > Tel.: +49(0)211 / 618 57 53 > Fax: +49(0)211 / 618 57 54 > > > -- > _ > -- 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 hangup request, cause 18
Ist the same operator connected to the pri-line? Perhaps another telco-operator can not connect to the desired destination - for whatever reason. Am 08.06.2011 12:55, schrieb Satish Patel: Thanks for reply, But I'm able to call those number from my cell phone and othere pri. I'm only having this issue on 2 pri line rest are working ? -- Sent from my iPhone On Jun 8, 2011, at 5:44 AM, Doug Lytle wrote: satish patel wrote: We are getting hangup cause 18 http://networking.ringofsaturn.com/Routers/isdncausecodes.php *Cause No. 18 - no user responding.* This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated. What it means: The equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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 hangup request, cause 18
Ist the same operator connected to the pri-line? Perhaps another telco-operator can not connect to the desired destination - for whatever reason. Am 08.06.2011 12:55, schrieb Satish Patel: Thanks for reply, But I'm able to call those number from my cell phone and othere pri. I'm only having this issue on 2 pri line rest are working ? -- Sent from my iPhone On Jun 8, 2011, at 5:44 AM, Doug Lytle wrote: satish patel wrote: We are getting hangup cause 18 http://networking.ringofsaturn.com/Routers/isdncausecodes.php *Cause No. 18 - no user responding.* This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated. What it means: The equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- -- _ -- 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: BYE is received late
For the record, it seems to be a SIP-ALG issue. It's fixed now. Vieri --- On Wed, 6/8/11, Vieri wrote: > Hi, > > I'm having an issue with all my calls going out my SIP > provider. I'm using > a softphone registering to a local Asterisk PBX (I'm using > Jitsi by the way - it's great and actively growing). > > I register as extension 4053 to asterisk server at > 10.215.147.115 (alias IP - > real IP addr. is 10.215.147.111) and dial a phone number > that is routed via > an Internet SIP provider. > The call is correctly established and conversation is OK. > If the local softphone user > hangs up first, the remote end is also disconnected > immediately. > However, if the remote party hangs up first, the local > caller is not > immediately disconnected. > That, of course, is undesirable. > > I'd like to understand why the call isn't automatically > hung up and fix it. > > I'm supposing that Jitsi isn't receiving a BYE as expected > in a correct SIP > transaction (or BYE is arriving very late). > I don't know why though. > > Here's my network setup: > > Softphone asterisk extension 4053 at 10.215.144.48 > Asterisk eth0: 10.215.147.111 but softphone registers to > the alias/floating IP > for failover setup 10.215.147.115 > Asterisk eth1: 192.168.103.111 > Asterisk default gateway: 192.168.103.1 > -> Asterisk accesses Internet via eth1 (192.168.103.1 is > a DSL modem/router) > > I did a tcpdump on the asterisk server while calling from > the local softphone as so: > tcpdump -s0 -X -n -w asterisk.cap -i eth0 host > 10.215.144.48 > > It's here: > http://213.96.91.201/temp/jitsi_via_asterisk.cap.gz > > Here's the full session (softphone waits 2 minutes until it > finally hangs up): > http://213.96.91.201/temp/jitsi_via_asterisk_full_session.cap.gz > > Asterisk seems to send BYE to the softphone after 120 > seconds since the remote party actually hung up... > > A packet dump on eth1 during the call also shows the BYE > message coming in from the SIP provider: > > http://213.96.91.201/temp/asterisk_eth1.txt > > I'm almost certain the remote SIP provider sends BYE in > time because earlier > today I tested by connecting the softphone directly to the > SIP provider and going out > the same DSL line (thus removing Asterisk from the > equation). ie. I placed a laptop with Jitsi in the same > subnet > 192.168.103.0 and used the default gateway 192.168.103.1 > (just like > Asterisk). All went well. > I also setup my Jitsi laptop within the 10.215.0.0 subnet > (just like my > Asterisk client setup) but connected directly to the SIP > provider (without > going through Asterisk). In this case the call ended as > expected (OK). > So I guess that something's wrong with my Asterisk > configuration. Both my softphone and network configuration > *should* be OK. > > However, it may have something to do with my Asterisk > eth0/eth1 setup but I don't see what. > > Any ideas/suggestions? > > Thanks, > > Vieri > > > -- > _ > -- 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] how to know length of file in seconds
Hi, I am using CentOS 5.6 and I am getting error message In my case old command is find. On Wed, Jun 8, 2011 at 5:25 PM, Karsten Wemheuer wrote: > Hi, > > Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger: > > On 11-06-07 02:31 AM, virendra bhati wrote: > > > Hi List, > > > > > > Is there any way by which we can get the length of any recorded files > into > > > seconds ? > > > > > > > $ sox foo.wav -e stat > > just a remark for people using newer(?)/other version of sox: In version > v14.3.0 (ubunto lyquid lynx) or v14.3.1 (Debian Squeeze) the above > command results in an error. You can use >sox foo.wav --null stat > instead. > > Karsten > > > > > -- > _ > -- 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 > -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] how to know length of file in seconds
Hi, Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger: > On 11-06-07 02:31 AM, virendra bhati wrote: > > Hi List, > > > > Is there any way by which we can get the length of any recorded files into > > seconds ? > > > > $ sox foo.wav -e stat just a remark for people using newer(?)/other version of sox: In version v14.3.0 (ubunto lyquid lynx) or v14.3.1 (Debian Squeeze) the above command results in an error. You can use sox foo.wav --null stat instead. Karsten -- _ -- 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 hangup request, cause 18
Thanks for reply, But I'm able to call those number from my cell phone and othere pri. I'm only having this issue on 2 pri line rest are working ? -- Sent from my iPhone On Jun 8, 2011, at 5:44 AM, Doug Lytle wrote: satish patel wrote: We are getting hangup cause 18 http://networking.ringofsaturn.com/Routers/isdncausecodes.php *Cause No. 18 - no user responding.* This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated. What it means: The equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- 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] Call queues on load-balanced asterisks
Hi Pan & Dhaval, In the past 8 weeks, we have delivered a load-balanced asterisks (1.4) based call center with our flexqueue application for icson.com. It has the below features, 1. 2 x asterisk 1.4 boxes, 1 x mysql db box and 1 x flexqueue box(the two are failover configured with heartbeat and custom script, and mysql master-slave replication between two svr), 2 x kamailio boxes(failover configured), 1 x file server boxes, 1 x app server , run freepbx & queuemetrics. all 8 server are dell r310. 2. the gateway is one mx100 with 4 E1 lines plugged, the incoming calls go to kamailio2 , and routed to ast1/ast2 in round robin mode. 3. all agent phones registered to kamailio 1, and the extensions are still maintained with freepbx 4.On asterisks, all trunks with destination to pstn or agent phones, go to kamailio1; and incoming calls trunk from kamailio2. 5.admin also use freepbx to configure inbound routes, ivrs, announcements, timeconditions, and recordings , etc. the configuration files are generated on the fly for flexqueue when apply changes. Dialplans for inbound routes are also automatically generated and distributed to ast1 & ast2, in these dialplan, fastagi application is installed as well to point to flexqueue. 6.flexqueue interprets the call flow configured on freepbx, and create the queues configured on freepbx, but it's shared among all asterisk boxes. 7.flexqueue interface with queuemetrics , and send all necessary queue logs to queuemetrics for complete reporting & QA purpose. 8.flexqueue has a agent phone panel, and a supervisor monitoring & management panel. Agent can logon his/her phone panel to have features like, incoming call popup, parking, Outbound dial, hold/unhold, transfer (cold/warm/to another queue), hangup, wrapup , pause/resume, etc. the supervisor can logon his/her monitoring & management panel, to view realtime event-driven agent info, queue info, and calls on-going. Besides, supervisor can also listen to agents, barge agents' talk, and qc call records & recordings quickly. 9.flexqueue provide web api for customer's CRM, which is asp.net based, to make agent can click-dial in their web crm application, and playback recordings to the agent's phone by clicking playback button beside crm communication records. The above system has been put into production from today, it's fully load-balanced asterisks based call queues or call centers. the gateways , the asterisk boxes can be added/removed any time. The fault asterisk box will be detected, and bypassed from routing destinations. I wish it's a good reference for your guys who want to create the same infrastructures. Best Regards, Thomas Liu - WShuttle Infotech Ltd. http://www.wshuttle.com / http://www.lookmypc.com http://www.vicidial.cn / http://www.call-center-software.com.cn Tel: +86 20 39230098 39230096 Mobile : +86 1390 3051 930 HK DID: +852 6950 0916, Macau DID: +853 6285 0645 Email: thomas@wshuttle.com MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly Yahoo Messenger: thomaslly Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, Guangzhou Higher Education Mega Center, Guangzhou, Guangdong Province, China. Zip code: 510006 -- > -Original Message- > From: Thomas Liu [mailto:thomas@wshuttle.com] > Sent: Wednesday, January 12, 2011 12:15 AM > To: 'asterisk-users@lists.digium.com' > Subject: RE: Call queues on load-balanced asterisks > > Hi Pan & Dhaval, > > We have implemented a FastAGI based queue with Erlang for a inbound call > center, and call this new application as FlexQueue. > All calls distributed on multiple asterisk boxes go through and are controlled by > that same remote fastagi server. > > It can routing calls to any destination, by any business rules. It don't rely on the > db for agent/call status store & query. > It's event driven and dict based agent/call store & query, with very good > performance, and low cpu power consumption. > > I think for your requirement, app_queue could not fulfill that. > > Best Regards, > > Thomas Liu > - > WShuttle Infotech Ltd. http://www.wshuttle.com / > http://www.lookmypc.com > http://www.vicidial.cn / http://www.call-center-software.com.cn > Tel: +86 20 39230098 39230096 > Mobile : +86 1390 3051 930 > HK DID: +852 6950 0916, Macau DID: +853 6285 0645 > Email: thomas@wshuttle.com > MSN:thomas...@21cn.com, QQ: 332148339, Skype:tonylly > Yahoo Messenger: thomaslly > Address: Room# 302, Building T8, Dongmen Plaza, Shuishi Reserved Area, > Guangzhou Higher Education Mega Center, Guangzhou, > Guangdong Province, China. Zip code: 510006 > --
Re: [asterisk-users] PRI hangup request, cause 18
satish patel wrote: We are getting hangup cause 18 http://networking.ringofsaturn.com/Routers/isdncausecodes.php *Cause No. 18 - no user responding.* This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated. What it means: The equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- 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] How to get DTMF in Konference module in Asterisk
HI Krishna, As per your suggestion I have changed Makefile of appKonference. Which is listed below. And after that I have reinstalled same module again. # turn app_konference dtmf on of off ( 0 == OFF, 1 == ON ) DTMF = 1 Now* how I know that DTMF is activated and working ? Is these any option by which we start it and save into any variable ? * I read on voip-info.org site about all options of konference where I got . *DTMF options:* 'X' : enable DTMF switch: video can be switched by users using DTMF. Do not use with 'S'. 'R' : enable DTMF relay: DTMF tones generate a manager event If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all members in the conference *So I use R option ? *Please guide me *how to save the DTMF into variable in dialplan when konference is going on ? * In the meanwhile I am also trying to get solution On Tue, Jun 7, 2011 at 9:29 PM, Krishna Sumanth Chava wrote: > Hi Virendra, > > Set DTMF option in the Makefile to "1" and then recompile/install the > app_konference module. > > Thanks > Krishna > > On Tue, Jun 7, 2011 at 1:31 AM, virendra bhati wrote: > >> Hi List, >> >> I am trying to get DTMF into conference room. for conference I am using >> Konference module. Konference don't have an option of DTMF gets. Is there >> any way by which I can get DTMF within conference room? >> >> >> >> >> - >> Thanks and regards >> >> Virendra Bhati >> +91-9172341457 >> Asterisk Engineer >> >> >> -- >> _ >> -- 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 > -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] how to know length of file in seconds
Thanks Paul, Link was too awesome. I read and check all related command too. Thank you for your help. On Wed, Jun 8, 2011 at 2:37 AM, Paul Belanger wrote: > On 11-06-07 02:31 AM, virendra bhati wrote: > >> Hi List, >> >> Is there any way by which we can get the length of any recorded files into >> seconds ? >> >> > $ sox foo.wav -e stat > > [1] - > http://www.thegeekstuff.com/2009/05/sound-exchange-sox-15-examples-to-manipulate-audio-files/ > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://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 > -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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 log in MySQL DB
Give it a shot and check! :) Yes you will have your Queue log records in table. [SATISH] On Wed, Jun 8, 2011 at 12:46 PM, Jonas Kellens wrote: > On 06/08/2011 09:10 AM, Satish Barot wrote: > >> >> Set queue_log = no in logger.conf. By default it is set to 'yes'. >> >> [SATISH] >> > > Will there then still be queue logging at all ? > > > > Kind regards, > Jonas. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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 log in MySQL DB
On 06/08/2011 09:10 AM, Satish Barot wrote: Set queue_log = no in logger.conf. By default it is set to 'yes'. [SATISH] Will there then still be queue logging at all ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue log in MySQL DB
Set queue_log = no in logger.conf. By default it is set to 'yes'. [SATISH] On Wed, Jun 8, 2011 at 12:30 PM, Jonas Kellens wrote: > Hello list, > > I have configured extconfig.conf to save queue log into my MySQL-DB. > > I notice however that there is still logging too in > /var/log/asterisk/queue_log. > > How can I disable queue logging into text files ? > > > > Kind regards, > Jonas. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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] Hints problem - NAT problem?
Hi all, I try to figure out why I have empty : > sip show subscriptions list in may asterisk 1.6. When device is registering to asterisk I can see in log: NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1010 but > sip show subscriptions is just empty. May it be the problem because devices are registering to asterisk from behind NAT? I belive this is the cause why hints does not work in my dialplan. Regards, Jarek -- _ -- 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] Queue log in MySQL DB
Hello list, I have configured extconfig.conf to save queue log into my MySQL-DB. I notice however that there is still logging too in /var/log/asterisk/queue_log. How can I disable queue logging into text files ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
