[asterisk-users] One-Way Calling between two * boxes (that was working before)
I have a site-to-site hardware (Cisco ASA-series on both ends) VPN between my house and our headquarters so I can occasionally telecommute and deal with personal calls in the office or work calls from home. I have an Asterisk install at home [11.6.0-rc2] with a SIP connection to our HQ Asterisk box [11.7.0] and a unified dialplan between the two sites (we have a remote office location with similar hardware and software configurations, also talking to the HQ site and part of the same dialplan that has working without issue). This has been working perfectly, but at some point in the past couple weeks I can't call from home to the office - calling from the office to home still works perfectly, as do calls to/from my landline. If I call the office voicemail extension and watch the Asterisk console in SSH, it's answering the call and playing back the prompts but as far as the house is concerned, the call is never answered and after several seconds the SIP trunk is "circuit-busy" and auto fallthrough as "CONGESTION" As far as I'm aware we aren't doing any traffic shaping/filtering/blocking on either end of the VPN. From home I'm able to SSH to LAN IP of the office * box, and from the office I'm a able to SSH to the LAN IP of my home * box. Running Packet Tracer on either ASA says that it's passing traffic on port 5060 in both directions . All of the Googling I've done turns up results relating to to one-way audio -- but that's not my problem here -- if the office calls me at home, I have no problems with audio quality in either direction. I'm just not able to complete a call from home to the office. Can anyone point me in the direction of what I need to poke at with a sharp stick? Thanks in advance! Lincoln -- Lincoln King-Cliby, CTS, DMC-E-4K/T/D Commercial Market Director Sr. Systems Architect | Crestron Certified Master Programmer (Gold) ControlWorks Consulting, LLC Crestron Services Provider | Biamp Audia Certified | Extron Control Professional -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Dialing a SIP URI with an ";ext=" parameter
Ok, so now I just feel kind of stupid. After I got home I decided to play with
this a little more.
After far too long I realized that part of the issue was Asterisk parsing the ;
as a beginning of a comment (hindsight=duh).
A little bit more experimenting and (though I could swear I tried this before)
replacing the ; with \; works.
That is, to dial a E.164 normalized number with an extension configured as
tel:+14404491100;ext=1407 with the SIP Peer for the Lync mediation server named
"lync" the working dial() is
Dial(SIP/lync/+14404491100\;ext=1407)
Hope this may save someone else time down the road.
--
Lincoln King-Cliby, CTS, DMC-D
Commercial Market Director
Sr. Systems Architect | Crestron Certified Master Programmer (Silver)
V: 440.449.1100 x1107 F: 440-449-1106 I:
http://www.controlworks.com<http://www.controlworks.com/>
Crestron Services Provider
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln King-Cliby
Sent: Monday, January 20, 2014 5:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dialing a SIP URI with an ";ext=" parameter
Hi All,
In the midst of trying to pilot a deployment of Microsoft Lync (mainly for
non-voice collaboration, specifically IM) and integrate it with our Asterisk
(11.6.0 if it matters) deployment and a "everything in one place" tool when
people are out of the office.
I have everything on the voice side playing nice from the Lync side
(Lync->Lync, Lync->Asterisk, Lync->Asterisk->PSTN) but I can't get calls from
Asterisk->Lync passing.
I think the root issue is Lync demands that the "line URI" be entered in a
E.164 normalized format, and further specifies that if an extension is
specified it should be entered as ;ext=. So, e.g. when I have myself set up in
LYNC my Line URI is entered as "tel:+144044911100;ext=1407".
If I try feeding that into an Asterisk DIAL() using any format I can think of
(specific examples below) the call fails and the following is logged to
console; it looks like Asterisk is dropping the ";ext="...
== Using SIP RTP CoS mark 5
-- Executing [1407@yyy:1] Dial("xx", "SIP/lync/"+14404491100")
in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/lync/+14404491100
-- Got SIP response 485 "Ambiguous" back from
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel ' xx' status is 'CHANUNAVAIL'
On the other hand, if I change my line URI to a "random" and unused in Lync
E.164 number without an extension and change the DIAL() to reflect that
number... the call succeeds, so it seems like I've narrowed it down to just
needing to figure out how to properly pass the extension to Lync.
The Googling I turned up didn't seem too positive (and suggested using an
Exchange Unified Messaging auto attendant and forcing the user to redial the
extension once connected to the AA was the only alternative for non-DID users)
but it seems like it should be relatively simple to bridge (what seems like a
very small) gap.
Here are the least embarrassing variations on Dial I've tried
Dial(SIP/lync/+14404491100;ext=1407) <-- 485 Ambiguous response as above
Dial(SIP/lync/"+14404491100;ext=1407") <-- 485 Ambiguous response as above
Dial("SIP/lync/+14404491100;ext=1407") <-- 485 Ambiguous response as above
Dial(SIP/lync/+14404491100/1407) <-- call 'sits there' and multiple "sip_xmit
of 0x7ffab40891e0 (len 841) to 0.0.5.127:5060 returned -1: Invalid argument"
logged to console
Any assistance, is as always very appreciated.
Thanks!
Lincoln
--
Lincoln King-Cliby, CTS, DMC-D
Commercial Market Director
Sr. Systems Architect | Crestron Certified Master Programmer (Silver)
V: 440.449.1100 x1107 F: 440-449-1106 I:
http://www.controlworks.com<http://www.controlworks.com/>
Crestron Services Provider
--
_
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[asterisk-users] Dialing a SIP URI with an ";ext=" parameter
Hi All,
In the midst of trying to pilot a deployment of Microsoft Lync (mainly for
non-voice collaboration, specifically IM) and integrate it with our Asterisk
(11.6.0 if it matters) deployment and a "everything in one place" tool when
people are out of the office.
I have everything on the voice side playing nice from the Lync side
(Lync->Lync, Lync->Asterisk, Lync->Asterisk->PSTN) but I can't get calls from
Asterisk->Lync passing.
I think the root issue is Lync demands that the "line URI" be entered in a
E.164 normalized format, and further specifies that if an extension is
specified it should be entered as ;ext=. So, e.g. when I have myself set up in
LYNC my Line URI is entered as "tel:+144044911100;ext=1407".
If I try feeding that into an Asterisk DIAL() using any format I can think of
(specific examples below) the call fails and the following is logged to
console; it looks like Asterisk is dropping the ";ext="...
== Using SIP RTP CoS mark 5
-- Executing [1407@yyy:1] Dial("xx", "SIP/lync/"+14404491100")
in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/lync/+14404491100
-- Got SIP response 485 "Ambiguous" back from
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel ' xx' status is 'CHANUNAVAIL'
On the other hand, if I change my line URI to a "random" and unused in Lync
E.164 number without an extension and change the DIAL() to reflect that
number... the call succeeds, so it seems like I've narrowed it down to just
needing to figure out how to properly pass the extension to Lync.
The Googling I turned up didn't seem too positive (and suggested using an
Exchange Unified Messaging auto attendant and forcing the user to redial the
extension once connected to the AA was the only alternative for non-DID users)
but it seems like it should be relatively simple to bridge (what seems like a
very small) gap.
Here are the least embarrassing variations on Dial I've tried
Dial(SIP/lync/+14404491100;ext=1407) <-- 485 Ambiguous response as above
Dial(SIP/lync/"+14404491100;ext=1407") <-- 485 Ambiguous response as above
Dial("SIP/lync/+14404491100;ext=1407") <-- 485 Ambiguous response as above
Dial(SIP/lync/+14404491100/1407) <-- call 'sits there' and multiple "sip_xmit
of 0x7ffab40891e0 (len 841) to 0.0.5.127:5060 returned -1: Invalid argument"
logged to console
Any assistance, is as always very appreciated.
Thanks!
Lincoln
--
Lincoln King-Cliby, CTS, DMC-D
Commercial Market Director
Sr. Systems Architect | Crestron Certified Master Programmer (Silver)
V: 440.449.1100 x1107 F: 440-449-1106 I:
http://www.controlworks.com<http://www.controlworks.com/>
Crestron Services Provider
--
_
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http://www.asterisk.org/hello
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To UNSUBSCRIBE or update options visit:
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[asterisk-users] Convert Asterisk Appliance (AA50) to "Open" Asterisk?
Hi All, Thanks for all of the help I've been given in the past and info I've picked up from this list over the years. I have an "official" Asterisk appliance (the AA50) running my PBX at home (we previously also had an AA50 in a satellite office-that one was recently retired and replaced with Asterisk running on commodity server hardware). Anyway - the AA50 software/Asterisk version is beyond outdated at this point, and the GUI has done nothing but infuriate me. Has anyone - or does anyone know if it's possible to - replace the "commercial" Linux/Asterisk running on the AA50 with another Linux flavor (say Ubuntu) and current open source Asterisk (ideally 11.something with Gareth's Cisco patch). I don't need - or want - a pretty GUI... just something I can SSH into and perhaps manhandle config files with Nano or something similar - worst case, something I can FTP/TFTP configuration files to. If that isn't feasible, anything low power/low profile/low cost that's particularly popular these days [bonus points if it's wall mountable/about the same size as the AA50]? My demands really aren't that severe -- one FXO, two FXS, a SIP "trunk" to the office (via hardware VPN), and maybe a half dozen Cisco 79xx phones. If it's not already apparent, I'm a relative Linux newb, but I'm farly well versed in patching and building Asterisk from source and generally getting things done once I have the pointers. Thanks in advance -- and happy new year! -- Lincoln King-Cliby, CTS, DMC-D Commercial Market Director Sr. Systems Architect | Crestron Certified Master Programmer (Silver) ControlWorks Consulting, LLC Crestron Services Provider http://www.controlworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Presence across two servers
Hi All, We've been running Asterisk for years in our offices but just recently replaced an Asterisk Appliance* in our smaller office with an actual server, upgraded the server in hardware in our HQ location and upgrading both ends to 11.5.0 with Gareth's patch for Cisco phones. 99.99% of our endpoints are Cisco 7961Gs. Each office is more-or-less standalone for ease of management and fault tolerance but we have a unified dialplan and SIP "trunking" from site to site via our VPN. Everything presence related works wonderfully for local users, but I'm hoping there's a way we could get presence for the people "at the other end of the pipe" fairly transparently. We have a lot of cross-office collaboration, and our office manager/receptionist (who has the battleship of a 7961G+7914+7914 BLF) would love to "at a glance" know if the remote folks are available for a call or not. I'm sure this has been covered, but my Googlefu us turning up a ton of redundant, old, and deprecated information so I've resorted to asking here. >From what I have found it sounds like it may be "easier" with IAX2 but my >experiments with IAX2 haven't yielded wonderful results and management prefers >"SIP everywhere" If anyone has any pointers I'd greatly appreciate it - thanks in advance! Lincoln *- One of the worst IT decisions I've made for better or worse. Looked good on paper; in practice not a good idea for anything beyond a very simple SOHO. -- Lincoln King-Cliby, CTS, DMC-D, CCMP-S Commercial Market Director Sr. Systems Architect | Crestron Certified Master Programmer (Silver) V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com<http://www.controlworks.com/> Crestron Services Provider -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Ring Two Extensions Simultaneously with different caller ID values? [SOLVED]
Luki,
Thanks for the quick response -- that did exactly what I was looking for and
was even easier than I thought.
Lincoln
--
Lincoln King-Cliby, CTS
Applications Engineer | Crestron Certified Programmer (Silver)
ControlWorks Consulting, LLC
V: 440.449.1100 x1107 | F: 440.449.1106 | I:http://www.controlworks.com/
Crestron Authorized Independent Progammer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luki
Sent: Tuesday, April 13, 2010 10:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ring Two Extensions Simultaneously with different
caller ID values?
Lincoln,
> Is there any way to feed different caller ID information to both sets while
> keeping them ringing simultaneously? My idea is to prefix the called
> extension to the name field (so as not to break redial/callback features on
> the phones)
you can do this with a local channel, like:
Set(__TARGET=${EXTEN})
Dial(SIP/phone1&Local/pho...@common_area)
[common_area]
exten => _phone.,1,Set(CALLERID(name)=${TARGET}: ${CALLERID(name)})
exten => _phone.,n,Dial(SIP/${EXTEN})
Something like that. I hope I got all the () and {} right, I don't do
that much dial-plan coding anymore...
Luki
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[asterisk-users] Ring Two Extensions Simultaneously with different caller ID values?
Hi All, We're using Asterisk 1.4, and Cisco phones exclusively (mostly the 7961G, but a few 7911Gs and one 7912G for the time being-all running the SIP firmware image, plus a few analog extensions until the next capital funding cycle). Each user has a phone at his or her desk, but there are also a growing number of "common area" phones (hallway, kitchen, conference rooms, data center, etc), and our users tend to be relatively mobile - revolving around their office, but might have a quick meeting in a conference room, or be BSing in the kitchen, or what have you. The request has come for users to know that their desk phone is ringing and be able to grab their calls, without having to log in/log out/forward their extension (after all a lot of this is very impromptu - you might just be waking back to the kitchen to grab a soda, bump into a coworker, and have a 20 minute discussion about widget X) It is, of course, quite easy to make multiple sets ring simultaneously (e.g SIP/&SIP/) - but both sets get the same caller ID info, so if I were to do this no one at the common area sets would know who the call was for. Is there any way to feed different caller ID information to both sets while keeping them ringing simultaneously? My idea is to prefix the called extension to the name field (so as not to break redial/callback features on the phones) For example, call from John Smith (x2789) to Jane Doe (x1234) would ring Jane's desk, the nearest conference room, and kitchen phones and present the caller ID as "John Smith" in her office, but "1234>John Smith" or something similar on the common area phones. Regardless of the advisability of implementing this in real life, does this question make sense and is there an easyish way to implement this? TIA for any pointers - and please keep in mind that I'm still a relative newbie for both Linux and Asterisk. Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.449.1100 x1107 F: 440.449.1106 I:http://www.controlworks.com Crestron Authorized Independent Programmer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] iphone, skype and asterisk ...
If the app is running it rings like any other softphone... if the app isn't running calls get diverted as if the client wasn't available. For my organization's application (primarily returning calls when out of the office/on the road or calling a coworker to confirm some detail/information in areas with poor cellular coverage) incoming calls are largely unimportant so I haven't paid much attention on that front. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, March 30, 2009 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] iphone, skype and asterisk ... 2009/3/30 Lincoln King-Cliby mailto:linc...@controlworks.com>> There are a couple SIP clients in the App store; the first one I tried (WeePhone) supports STUN but breaks horribly when you try to connect to an Asterisk server across a VPN connection, which is critical in our case since we aren't exposing our Asterisk deployments to the public Internet (same reason why Fring won't work for us). On the same LAN as the Asterisk server with no intervening VPN it has a few quirks but seems to work decently enough. How are incoming calls working ? >From here (http://code.google.com/p/siphon/wiki/FAQ_en), I can read : "Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background. So, when Siphon doesn't run, the SIP server of your provider doesn't know your iPhone." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iphone, skype and asterisk ...
There are a couple SIP clients in the App store; the first one I tried (WeePhone) supports STUN but breaks horribly when you try to connect to an Asterisk server across a VPN connection, which is critical in our case since we aren't exposing our Asterisk deployments to the public Internet (same reason why Fring won't work for us). On the same LAN as the Asterisk server with no intervening VPN it has a few quirks but seems to work decently enough. It looks like the problem is that WeePhone registers with Asterisk using either the local LAN (WiFi) IP address or the public WAN IP address discovered using STUN [if enabled]; there doesn't seem to be a way to force it to use the VPN IP when registering, which, of course, makes it nearly useless for my application - but if you have an Asterisk server with the proper ports exposed to the public Internet that may not be a problem for you. Anyone know if this is true with the other SIP clients in the App Store? Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, March 30, 2009 7:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] iphone, skype and asterisk ... Hi, >From various readings, I thought that the main hurdle that kept iphones away >from asterisk were : 1. a clause in iphone Developpers agreement that forbid applications running in background, 2. lack of sip clients. Now it seems skype is available on iphones. Has someone tried it ? Along new skype capabilities in Asterisk, could it be used to hook iphones to Asterisk for both inbound and outbound calls ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No reply to our critical packet
My best guess at the root cause of the problem after looking at the packet capture was that the phone was not happy seeing the call "connected" before any of the intermediate states (trying, ringing, etc.) and Ringing() generated the session progress (e.g. in addition to the in-band ringback it also generates the SIP message to tell the phone that the phone is ringing... or maybe it just generates the SIP message and the phone generates the ringback) necessary to make the phone happy; I don't think Playback() does the same thing. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, March 13, 2009 2:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No reply to our critical packet Not a better hack but perhaps more palatable to the listener Playback(please-wait) Wait(1) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln King-Cliby Sent: Friday, March 13, 2009 1:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No reply to our critical packet I had this issue with SIP Phones (Cisco 7961) to local voicemail; the issue was resolved by adding a Ringing() followed by Wait(1) before the VoicemailMain() in the dial plan... it seems like there should be a better way, and I feel it's rather crude to force the user to listen to a second of ringback before launching into voicemail, but it solved the problem for me (and yes, I did try just Ringing() with no wait with no such luck) -- maybe it would work in your case with a SIP call as well? HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No reply to our critical packet Hi, thanks for the quick reply. > 1. Do you have the incoming 1-2 holes in your firewall so the > remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. > 2. If #1 is ok, try putting an Answer command in front of your Dial > Command. Doesn't help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No reply to our critical packet
I had this issue with SIP Phones (Cisco 7961) to local voicemail; the issue was resolved by adding a Ringing() followed by Wait(1) before the VoicemailMain() in the dial plan... it seems like there should be a better way, and I feel it's rather crude to force the user to listen to a second of ringback before launching into voicemail, but it solved the problem for me (and yes, I did try just Ringing() with no wait with no such luck) -- maybe it would work in your case with a SIP call as well? HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No reply to our critical packet Hi, thanks for the quick reply. > 1. Do you have the incoming 1-2 holes in your firewall so the > remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. > 2. If #1 is ok, try putting an Answer command in front of your Dial > Command. Doesn't help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware that can accomondate 2 TDM24
Are you locked into the 3U form factor? We're running Asterisk on a Dell PowerEdge 1950 (1U, 2 full height PCI-E slots [one home to an AEX-804E], 3 drive bays, redundant power). I both the 2950 and 2970 (both are 2U, variable number of drive bays based on the config you choose, the 2950 shares firmware with the 1950) can be ordered with PCI-E risers because we have a handful in our datacenter, but I have no idea how many slots -- I want to say 3. I think the TDM24 is too long to fit in a 1950, but I'm pretty sure (you'd have to check) that the 2950/70 has at least two full-length slots. HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 F: 440.729.0884 I:http://www.controlworks.com Crestron Authorized Independent Programmer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelvin Chan Sent: Wednesday, February 04, 2009 7:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] hardware that can accomondate 2 TDM24 Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. 604-585-2...@104 (main) | Surrey, BC 604-585-3056 (fax)| Canada, V3W 1R1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail
Hi Steve, Thanks again for the response-- the answer you gave was more or less the answer that I was expecting. I was logging all packets to and from the phone, and I never saw an ACK from the phone for the OK to Asterisk on the VM calls -- not an ACK directed to a different location, just no ACK period. I noted in my other reply that as a test I had added a call to Ringing() followed by Wait(1) before dropping into Voicemail for the voicemail extension in the dialplan, since I noticed that the only difference that appeared to exist between a SIP-POTS or SIP-SIP call and a SIP-Voicemail call, aside from the missing ACK from the phone is that Asterisk reported session progress of "100 Trying" and "180 Ringing" to the phone, where it didn't report either of these when calling Voicemail, instead jumping straight to "200 OK with session description". In the 24 hours since I did that we haven't been able to get any of dozens of calls to Voicemail to fail, when normally it would borderline on greater than one in every two call. I'm still not convinced it's fixed, but I'm feeling fairly good about the solution, so it seems to my untrained eye like there may be an issue in the Cisco 79x1 firmware if the PBX "accepts" a call without providing any intermediate status? That seems like it would manifest itself in other places, and I'm kind of grasping at straws but... Thanks again to everyone who took the time to read and or respond to this issue -- I'll post again if it turns out that that wasn't actually the fix, but for now management is happy that they can actually listen to their entire voicemail messages. Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 F: 440.729.0884 I:http://www.controlworks.com Crestron Authorized Independent Programmer -Original Message- From: Steven J. Douglas [mailto:stev...@moij.biz] Sent: Wednesday, February 04, 2009 12:43 AM To: Lincoln King-Cliby Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail Hi Lincoln, The fact that you can hear and respond to the voice mail (even if its for the first 20 seconds), means that your phone has received the OK message properly. The problem is the missing ACK after receiving OK. When asterisk did not receive the ACK after a few retries of the OK, it terminated the call. This resulted in your RTP streams getting the icmp error messages. Assuming that you are capturing every packet that goes on between Asterisk and the phone, there are two possibilities. 1. The phone has a bug. 2. The ACK was sent somewhere else. Normally the ACK message destination is constructed from the response to the INVITE. In this case, it will be the OK message. If you suspect its the second case, you can capture the traffic for both a good voicemail call and a failed voicemail call. Then by comparing the messages, you might get a hint. If you need help, you can send the packet capture to me privately (not through the list as it might be a large file) and I can help vet it for you. Unfortunately there is no flag that you can set to confirm a session based on OK being transmitted and not wait for ACK. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail
-Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Mark Wiater > Sent: Tuesday, February 03, 2009 2:25 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls > > Dropped in Voicemail > > Wouldn't this suggest that either Asterisk couldn't open the port, > or opened it and then closed it? Or I suppose that perhaps the phone > and asterisk didn't negotiate the port properly? Since the RTP packets had up until that point been flowing in both directions, I'm guessing it's the middle (opened it and then closed it); no change in ports in the log, and this would seem to correspond roughly to the point at which Asterisk "gives up" on retrying the packet > In your original post, I thought I read that you could reproduce > this issue by increasing load on the asterisk server. What does the > caller experience in the first 20 seconds when a call to voicemail > is going to fail? Just ringing? I guess I should have been a little bit more clear on that one: Asterisk always answers calls to Voicemail more or less instantaneously. Reproducing it has been very hit-or-miss with no real correlation to network activity, call volume, time of day, day of week, phase of moon, etc. However, more often than not you can force the issue to manifest itself by by making several "rapid-fire" calls to voicemail from the same phone. The first 20 seconds of a call that ultimately fails is indistinguishable from a call that doesn't fail - Comedian Mail answers, accepts DTMF input from users, starts playing back messages, etc., and then all of a sudden at about the 20 second mark the audio dies. The phone still thinks that it's in a call but no more audio gets sent to the phone, DTMF input is ignored, etc. > Any chance there's anything in Asterisk's or the OSes logs about > some failure of the network stack? What OS is this? This is Ubuntu Server 8.1.0; as far as logs go, please excuse me -- my background is primarily Windows so I'm not sure where I would find those (but I'll Google that now ;) ) Also, an interesting side note: While I don't want to call the issue "fixed" yet based on how intermittent it has been, since I noticed that Asterisk indicated "Ringing" to the phone on SIP-to-SIP calls (that have never failed) I added: Voicemail,2,Ringing() Voicemail,3,Wait(1) To the Voicemail extension in the dialplan (I literally use "Voicemail" as the extension that the Messages button on the phone dials), and so far I have not been able to reproduce my issue... I don't like it because the user hears a ring cycle before the VM attendant answers, but if it keeps them from being bounced out of VM in the middle of listening to a message, I can live with it. Anyone with more knowledge of the inner workings of things want to tell me if I should or shouldn't be surprised if the issue reappears with this in place? Thanks again for the help! Lincoln ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail
> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Steve J. Douglas > Sent: Tuesday, February 03, 2009 3:30 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls > > Dropped in Voicemail > > Hi Lincoln, > > Asterisk was expecting ACK after sending the 200 OK message. After > repeated attempts at sending the 200 OK message and not receiving ACK, > it terminated the call. Are you able to do a packet capture on the phone > end? Mostly likely the phone is sending the ACK, but its either sent to > somewhere else or your firewall is blocking it (not likely since you are > able to receive the call in the first place). The packet capture on the > phone end will probably show you the smoking gun. Hi Steve, as I noted in an earlier reply the * box and the phone are on the same switch, subnet, and VLAN -- there's no routing or firewall between the two. I enabled port mirroring on the switch for the port that the phone is plugged into and ran Wireshark while a call was placed. I'm not really 100% sure of what I -should- be seeing so I'm not sure if what I'm seeing is correct or not. The call progress that I'm seeing is starting at 16.550698 seconds into the capture: 1237 Phone -> Asterisk: INVITE sip:voicem...@10.2.0.2 1238 Asterisk -> Phone: 407 Proxy Authentication Required 1239 Phone -> Asterisk: ACK sip:voicem...@10.2.0.2 1240 Asterisk -> Phone: 100 Trying 1241 Asterisk -> Phone: 200 OK, with session description At that point a ton of RTP packets are exchanged with a couple ARP lookups from the phone asking for the Asterisk server. (packets 1245-1247 @ 16.6066 seconds, 1295-1297 @ 17.5701 seconds) Then there's 1299 Asterisk -> Phone: 200 OK, with session description (@ 17.520362 seconds) With more RTP packets (including a couple with DTMF payloads) followed by 1402 Asterisk -> Phone: 200 OK, with session description (@ 18.572025 seconds) More RTP packets and DTMF follows with 1607 Asterisk -> Phone: 200 OK, with session description (@ 20.570493 seconds) More RTP and at 21.570456 seconds there's a "RTCP" "Sender Report Source Description" from Asterisk to the phone, and at 21.745548 there's an NTP sync from the Phone to one of our network time servers. Much more RTP follows with 2011 Asterisk -> Phone: 200 OK, with session description (@ 24.573025 seconds) Much more RTP follows with one more "RTCP" "Sender Report Source Description" and then there's 2412 Asterisk -> Phone: 200 OK, with session description (@ 28.570687 seconds) --- you get the idea --- 2814 Asterisk -> Phone: 200 OK, with session description (@ 32.570784 seconds) Then starting at packet 3217 there are a series 6 of ICMP "Destination unreachable (Port Unreachable)" messages from the Asterisk server to the phone, with an RTP packet from the Phone to the Asterisk server before each Destination unreachable message. After that series the phone sends a series of 44 RTP packets to the * Box without getting anything back. There's then a lone ICMP Destination Unreachable (Port Unreachable) message. This sequence repeats about 10 times until the user hangs up, at which point: 3721 Phone -> Asterisk: BYE sip:voicem...@10.2.0.2 (@46.335605 seconds) 3722 Asterisk -> Phone: Status: 481 Call/leg transaction does not exist Now, on the other hand, a phone-to-phone call (actually the user calling my phone to let me know it failed), I do see an ACK very early on That process is Phone->Asterisk Invite Asterisk->Phone 407 Proxy Authentication Required Phone->Asterisk ACK Phone->Asterisk Invite Asterisk->Phone 100 Trying Asterisk->Phone 180 Ringing Asterisk->Phone 200 OK with session description (RTP Packets are exchanged) Phone->Asterisk ACK (RTP Packets are exchanged) Any idea why the phone would be ACKing phone-to-phone calls but not ACKing phone-to-voicemail calls? Any way to make Asterisk not drop a call just because it wasn't ACKed even though packets are still going back and forth? Thanks again for any help! Lincoln ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail
Thanks to everyone who has replied so far; to answer a few of the follow up questions that have been posed: Dave - > Which firmware load? We had all kinds of trouble with 8.4.x, after being > stable for a few months on 8.3.x. Going back to 8.3.x made all of the > weirdness disappear. While we're on the Cisco note, I have script to > remotely reboot the SIP firmware load Ciscos and to provision the phones > based on active directory if you're interested... back on topic: Hmm... very interesting with regard to the script. For firmware I've tried everything from 8.3.5 up to 8.4.2 with no concrete change (there was one version of 8.3.x that "seemed" to be a little worse, but with how hit or miss it is who knows what reality was) The other thing that's been driving me crazy is we have an Asterisk Appliance in one of our remote offices, and while the Appliance is a PITA to admin, neither of the users over there have reported any issues with any of the FW images I've pushed out. > Have you run a packet cap on a mirror of the switchport the phone this is > happening on is connected to? Anything strange? What's happening on the > switch backplane (network backbone) at large when you notice the problems? > > Major transfers/lots of traffic? Anything else running on the * server? I'll need to set something up to do a packet cap; the network itself is relatively quiet, and I've been able to reproduce the issue after hours so user-generated traffic isn't in play. The * server also hosts Flash Operator Panel and Zaptel (4 FXO lines) but that's it. There are only 5.5 FTEs and 7 sets in this office so I can't imagine that we're putting that much stress on the box. Alex - > Sounds like there's some sort of firewall in place or something else that > is preventing an ACK from being received in response to the 200 OK. > Notice that the 200 OK keeps being retransmitted. Nope, Asterisk box and phones are on the same subnet and VLAN with no routing or firewalling between the two. In fact, the phone that experiences the issue most frequently is connected to the same physical switch as the Asterisk box. Steve- > I have a customer with the same complaint and I am trying to figure it out > as well. I have not caught the debug action yet though. > > First, are you using FreePBX? Second, are you using the "announce" > feature. No to the FreePBX question... Just plain ole regular Asterisk 1.4.22 built from source running on Ubuntu. Not sure how to answer the "announce" question -- when I searched voicemail.conf nothing was found. Thanks again! Lincoln ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail
1980 retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 - no reply to our critical packet (see doc/sip-retransmit.txt). Really destroying SIP dialog '001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203' Method: INVITE Sending to 10.2.0.203 : 5060 (no NAT) cworks-phones1*CLI> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK446b7de3;received=10.2.0.203 From: "Jim Felderman" ;tag=001d45b61d4906959ea33ab4-af2b7b8b To: ;tag=as53449c29 Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <> Scheduling destruction of SIP dialog '0d32ee8515d9f6dc4439002f1d601...@10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.2.0.203:5060: NOTIFY sip:1...@10.2.0.203:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK0cb71f67;rport From: "asterisk" ;tag=as0b88d5a9 To: Contact: Call-ID: 0d32ee8515d9f6dc4439002f1d601...@10.2.0.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 83 Messages-Waiting: yes Message-Account: sip:aster...@10.2.0.2 Voice-Message: 2/6 --- Really destroying SIP dialog '0d32ee8515d9f6dc4439002f1d601...@10.2.0.2' Method: NOTIFY cworks-phones1*CLI> -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Dropped in Voicemail / No Reply to Our Critical Packet w/ SIP Debug
60;branch=z9hG4bK0cb71f67;rport From: "asterisk" ;tag=as0b88d5a9 To: Contact: Call-ID: 0d32ee8515d9f6dc4439002f1d601...@10.2.0.2 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 83 Messages-Waiting: yes Message-Account: sip:aster...@10.2.0.2 Voice-Message: 2/6 --- Really destroying SIP dialog '0d32ee8515d9f6dc4439002f1d601...@10.2.0.2' Method: NOTIFY cworks-phones1*CLI> -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Appliance
From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Pierce [pier...@westmancom.com] Sent: Tuesday, January 13, 2009 5:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Appliance > > I'm looking for some info on the Asterisk Appliance. > > I understand it has a gui, but can I still do all the dialplan config > that I'm used of doing by hand outside of the gui? If I really wanted > to, could I even ignore that the device has a gui and do all my config > in the files? I guess I'm just wondering if it will be as flexible as a > 'vanilla' asterisk install from source on a linux system. > > Also, from those who are using these devices, what has your experience > been? Are they stable? Do they seem to have enough horsepower and > storage space for an SMB with up to 50 phones? Some older specs stated > they would be appropriate for businesses with 2-50 users, while the > current spec on the Digium site states they are appropriate for 2-20 > users. Sorry for the delay in responding -- I'm just now having time to catch up on a month of the list, but it doesn't appear anyone else has responded. We have an Asterisk Appliance in a remote office (and I have one at home) and it's a mistake we would not make again. Concept wise it's a nice idea, but using Asterisk built from source is so much easier, more flexible, and less stressful. I guess our biggest issue with the appliance isn't hardware but administrative: While I found no reference to it before we purchased the Appliances (maybe I didn't look hard enough, maybe it's not documented anywere that's readily findable) Digium doesn't support any confiuguration not generated using the GUI "wizard"... Digium doesn't support FTPing files onto or off of the appliance, Digium doesn't support... Stability wise I don't have any complaints, it's never crashed on us except when I tried FTPing config files off so I could edit them. We only have two full-time users at the remote office, and I have only me at home (6 total extentions at the office for visiting staff, 7 extensions at home). There is a wierd FXO caller ID issue that we're fighting on one of the POTS lines, but Digium essentially refused to support it because we aren't running anything close to what the GUI could build for us "automagically" (unified dialplan across three sites, either site can call out on the other site's POTS lines, call queuing that crosses sites, feature codes to force calls to extensions direct-to-voicemail (no ring ever) and ring-for-eternity (no voicemail ever, etc.) [in fairness it looks like its a CO issue because the problem follows the line, the other POTS lines don't exhibit the issue, though all lines work with a "standard" caller ID box]. I'm weary of making changes to that box because while you can edit config files through the GUI, if you aren't very careful it seems like parts of a context will rearange itself (e.g. part of extension "s" will end up in the middle of something completely different). We tried to upgrade the firmware once and it screwed things up to the point where for a day the site had no telephone service (oh, suprise! If you have any custom stuff in extension.conf and upgrade the firmware it will spin off into an infinite loop orbit until you factory default the thing... and once we got control of the infinite loop issue we couldn't get the config that had been working just beautifully to work at all, so we punted and rolled back to the previous FW version) Basically, my experience with the appliance it it's a beautiful little box and if they'd just ditch the GUI and give a moderately user friendly commandline text editor I'd be on it in a second, but with the appliance in it's current state it took me less time to * Unbox a Dell PowerEdge 1950 * Install AEX804E card * Rack a Dell PowerEdge 1950 * Install "server version" of Linux from Distro CD * Download Aserisk and Zaptel sources, compile, configure, and install * Build Dialplan by hand from scratch (and I think, not including labor costs this was also close to the same cost, if not cheaper than the appliance) Then the amount of time I've spent trying to get the Appliance to do what we want -- and what we want for the thing really isn't that complicated (really just "Take POTS lines in, handle local switching for a handful of extensions so that that traffic doesn't wind up on our WAN/VPN connecions, and act as a SSU"-- we aren't even doing queues on it!). Plus making changes to the Dell version of things is also quicker and less nerve wracking.. As you can probably tell I don't care much for the Appliance based on our understanding of what it was. On the other hand, though, if all you have is one site and just need a basic SOHO PBX it's a decent contender... the support more than anything is what's left a ba
Re: [asterisk-users] Possible to get "Courtesy Tone" on attended transfer?
Thanks for the answer Terry, it's kind of what I expected. I may have to look into using Attended transfers in Asterisk, but I think my users really prefer having the TRNSFR soft key instead of remembering a feature code. I guess then the next question... Does anyone know of a way to map the Transfer key on a Cisco 79x1 (Specifically 7961G) to call the Asterisk attended transfer function vs. doing a native SIP transfer? -- As I think of it, even if it were possible, that may create more issues than it solves, but I'd still be curious if there was a way to do that. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry Wilson Sent: Friday, December 05, 2008 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Possible to get "Courtesy Tone" on attended transfer? > Is there any way to provide the user receiving an attended transfer > with a tone or other audible indication that the transfer is > completed (i.e. Party A calls Party B, Party B announces the call > while transferring to Party C, Party C hears tone when Party B > completes the transfer so that they know that they are now talking > to Party A instead of Party B)? > If you are using builtin asterisk attended transfers (enabled in features.conf), then this is the behavior. I don't believe there is a way to do this with native SIP transfers, currently. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Possible to get "Courtesy Tone" on attended transfer?
Hi All, Is there any way to provide the user receiving an attended transfer with a tone or other audible indication that the transfer is completed (i.e. Party A calls Party B, Party B announces the call while transferring to Party C, Party C hears tone when Party B completes the transfer so that they know that they are now talking to Party A instead of Party B)? I know this is possible when picking up a parked call, but I haven't found any information re: transferred calls. My users are finding that they don't know when to begin talking to the person on the other end of the call. Similarly, but less important, is there any way to push the original caller ID over to the extension that receives the transferred call? Thanks in advance, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip MWI Messages-Waiting: always reports no messages
In sip.conf do you have [EMAIL PROTECTED] Lincoln From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Mark G. Thomas [EMAIL PROTECTED] Sent: Wednesday, November 26, 2008 11:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip MWI Messages-Waiting: always reports no messages Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like "[EMAIL PROTECTED]" in sip.conf, but that doesn't help any. I also tried the same with the mailbox containing messages under [default], but still no luck. I see messages like this if I do "sip set debug ip 10.5.7.130" - Reliably Transmitting (no NAT) to 10.5.7.130:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.7.21:5060;branch=z9hG4bK44627853;rport From: "asterisk" ;tag=as7d9b65d4 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 84 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) --- - from sip.conf: -- [29] insecure=port,invite context=ccme type=friend host=r2.home.misty.com qualify=yes dtmfmode=rfc2833 canreinvite=no nat=no mailbox=29 fromuser=777 vmexten=777 username=29 from voicemail.conf: --- [home] ; testing 29 => 1234,Joe Test,[EMAIL PROTECTED] --- [EMAIL PROTECTED] asterisk]# ls /var/spool/asterisk/voicemail/home/29/INBOX msg.gsm msg.WAV msg0001.wav msg0002.txt msg0003.gsm msg0003.WAV msg0004.wav msg.txt msg0001.gsm msg0001.WAV msg0002.wav msg0003.txt msg0004.gsm msg0004.WAV msg.wav msg0001.txt msg0002.gsm msg0002.WAV msg0003.wav msg0004.txt -- Mark G. Thomas ([EMAIL PROTECTED]) voice: 215-591-3695 http://mail-cleaner.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Low RX volume and half duplex/"walkie-talkie" on AEX-804E
Hi All, I have a ticket open with Digium, but based on their previous lack of support for the Asterisk Appliance, I'm not really holding my breath - and, honestly, I'm not 100% convinced it's a Digium issue in the first place (but I don't know where else to point fingers). We have an AEX-804E (PCI Express, 4 FXO ports, Hardware Echo Cancellation) in a Dell PowerEdge 1950 with four straight analog telephone lines, and running asterisk 1.4.22. All of the local phones are Cisco 7961G with the SIP firmware. Calls between SIP sets, across our SIP trunk on a VPN to a remote office, or calls to or from the remote office's PSTN lines (over the aforementioned SIP trunk) are all fine. On many [but not all] calls to or from the PSTN, I'm getting two complaints - #1 is low receive (i.e. from the PSTN) volume #2 (which seems to get significantly worse if I try tweaking bumping up the tx/rx gain in Zapata.conf) is that if the person in our office is talking all inbound audio is muted, but not the other way around (i.e. half duplex, but not half duplex both directions if that makes any sense) Further compromising my sanity is that #1 seems hard for me to duplicate - calls to or from my cell phone, for example, always sound fine. Local calls are "mostly" fine, and long distance calls are hit-or-miss, calls to a Hawaiian (how's that for "Long Distance" from Ohio) 1004 Hz test number are fine - in fact, subjectively, borderline too loud which makes no sense since before going live with Asterisk, we had a legacy Panasonic KSU/PBX on the same lines - on the same punchdown blocks - and no one ever complained about these issues. If I turn off the echo canceller there's a modest (may even just be psychological) improvement in line gain, but the echo is so horrendous (actually the echo sounds louder than the inbound call volume) as to make things unusable. Any ideas? At all? I'm still relatively new to the Asterisk-interconnected-to-PSTN side of things, and it seems like there are dozens of config files and tools so explicit instructions are appreciated! Thanks in advance, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iPhone SIP or IAX client (without proxy)?
I checked the app store and haven't found anything promising, but I figured I'd ask here. Does anyone know of a SIP or IAX client for a non-jailbroken iPhone that will communicate directly with a machine running Asterisk? I know that there's at least one offering that seems like it's essentially a proxy (App runs on iPhone, iPhone talks to 3rd party server, 3rd party server talks to SIP service), but that's not an option we consider viable. Our Asterisk servers have no IP exposure to the outside world (everything in and out is POTS)-and that's not going to change-but we're using Cisco Security Appliances with VPN access from the outside. The iPhone has Cisco VPN support "out of the box". Ideally, I'd like to get the phones that aren't already on the VPN on the VPN, configure an extension for the phone and give our folks (who split their time about 60% in the office 40% in the field) access to the corporate phone system from the field without having to fire up a laptop and pull out a headset. TIA, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.controlworks.com<http://www.controlworks.com/> Crestron Authorized Independent Programmer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP
Just as an interesting follow-up/additional information, if I place a call to Site 2 on a POTS line, someone at Site 2 answers the call (using one of the Cisco phones) and then transfers it to me across the VPN the call sounds fine. So I think Bob's question was on the right track with it being a CODEC issue, but I'm not sure how I need to deal with that for the ZAP channel type. Thanks again, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln King-Cliby Sent: Monday, November 03, 2008 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is "done for you" but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -v -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: > Any ideas why the audio quality would be so markedly different when > the only thing that seems to be different is where the call is > originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP
Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is "done for you" but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -v -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: > Any ideas why the audio quality would be so markedly different when > the only thing that seems to be different is where the call is > originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call quality issue across VPN-> POTS vs SIP
Hi All,
Got a strange (at least IMHO) issue that doesn't make much sense to me.
Basic configuration is two sites with a site-to-site (aka router-to-router)
VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks,
and the only VoIP is internal - all of our outward telecom is on POTS or
Centrex-enabled POTS lines.
Site 1 has a Dell PowerEdge 1950 with Asterisk built from source and an AEX804E
to connect to the outside world. Site 2 has an Asterisk Appliance with the 4
FXO / 4 FXS configuration, with the FXS ports currently unused.
The PBXes at each site are configured to be essentially independent but with a
unified dial plan so that calls can be placed or transferred across the VPN
with a SIP trunk connecting the two PBXes, and canreinvite=no is set
everywhere. The only other "heavy" consumer of bandwidth across the VPN is a
real-time file replication suite that we use for file synchronization. While
this is the ultimate issue, I don't understand the phenomena I'm seeing:
If a user dials in to one of Site 2 FXO lines then dials across the VPN to a
user at Site 1 while the file replication job is running audio quality (to the
caller on the POTS line only) is abysmal-- audibly it sounds like about 50% of
the packets are dropped ("He__ Th__ __u __r, T__s is ___ln")
On the other hand if a user at Site 2 picks up one of the Cisco phones [with
the replication job still running] and dials across the VPN to a user at Site 1
audio quality is fantastic, ditto if a user at Site 1 calls to a user at Site 2.
Any ideas why the audio quality would be so markedly different when the only
thing that seems to be different is where the call is originating from (POTS
line vs. SIP phone)?
Replacing the border gear with equipment that's QOS aware and can handle
prioritization is already on the list (and may be in the process of being
ordered at this point)
Thanks,
Lincoln
--
Lincoln King-Cliby, CTS
Applications Engineer
ControlWorks Consulting, LLC
Crestron Authorized Independent Programmer
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Re: [asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Benny and Mark, Thank you for your replies. I tried adding t1min=500 to sip.conf per the suggestion below and since doing that haven't been able to reproduce the issue. If it comes back, I'll do the SIP debug per Mark's suggestion and post the results here. (Mark, per your question the Auto Attendant and Voicemail are on the same box) Thanks again for the quick help! Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.thecontrolworks.com/ Crestron Authorized Independent Programmer -Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 28, 2008 5:20 PM To: Lincoln King-Cliby Cc: 'asterisk-users@lists.digium.com' Subject: Re: Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet In Asterisk 1.2.x it could sometimes happen that Asterisk believed the path to a server was so good, that it would only allow 1 ms for answers to be received. It would do all its retransmissions in less than 200ms, and then it would complain about no reply to critical packet. Anyway, you can adjust the minimum timer with the configuration option t1min in sip.conf. I would recommend setting it to at least 100 (it is in ms) and perhaps 500 would help for you. It is also highly possible that your issue is completely different. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] XML Cisco config file
I'm not sure if it's the only issue but you're going to have issues with Etiqueta_del_telefono The text within the tag is a maximum of 11 or 12 characters (I can't remember off the top of my head), if it's longer than that--I count 21 characters in the example, the phone will reject the entire configuration file more or less silently (it is logged in the phone's debug log at http:/// but there's no display on the phone itself). That sounds like at least part of what's happening in your case. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.controlworks.com<http://www.controlworks.com/> Crestron Authorized Independent Programmer From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of César García Sent: Tuesday, October 28, 2008 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] XML Cisco config file Hello guys, anybody here that can help me checking out this xml file, cause I am traying to configure some cisco 7911G phones to asterisk and I can't get it done thanks a paste of the file is here: http://pastebin.ca/1239083 -- http://celord.blogspot.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Hi All, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all phones are on the same subnet (10.2.0.x/255.255.255.0) of the local LAN with no NAT, routing, firewall, etc., etc. between the server and the phones. Periodically I'm seeing calls placed from the 7961s through anything on the PBX that requires digit entry (the Auto Attendant, Voicemail, etc.) 'randomly' drop; extension-to-extension calls extension-to-PSTN, and PSTN-to-extension calls never have any issues whatsoever. Nor have I been able to duplicate the issues hopping around auto attendants on an inbound PSTN call. When the call drops, the phone still thinks that it is connected, but the audio path is cut off and something similar to the following is dumped to the console -- Playing 'vm-password' (language 'en') [Oct 28 14:23:07] WARNING[9423]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 28 14:23:07] WARNING[9423]: chan_sip.c:1980 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). All of the results Google has turned up, and the doc/sip-retransmit.txt file point to problems with things in the middle of the path between the server and the phone (NAT, firewall, "SIP middle box", proxy) that simply don't exist in the configuration that we're using. I suspect it's an issue with the way the Cisco phones are dealing with DTMF to Asterisk or Asterisk dealing with the DTMF from Cisco but that's where I go off into unknown territory. (FWIW, until the call drops everything works fine, pressing a button triggers the desired action, and audio quality is fantastic) I've rolled the firmware on the phones up and down with no noticeable change, and I also upgraded to Asterisk 1.4.22 version of Asterisk (I had been running 1.4.21.2, and there are fewer dropped calls with .22 but it's still way too often to be acceptable) Any suggestions are greatly appreciated, but please be explicit... short of editing the configuration files and "make install" my Asterisk experience is rather limited. Thanks in advance, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC http://www.thecontrolworks.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
