Re: [asterisk-users] asterisk SIP, SIPAddHeader() and Cisco GED-125
Hello David, I had an application where I had to pass data between Asterisk and a Genesys system using SIPAddHeader(). It worked pretty well, but we had two genesys boxes, and by the time I was done I found I was losing the SIP header where I needed it, since it only shows up on next INVITE. I ended up storing data in the CallerID Name field with a delimeter and parsing it out. Far from an ideal solution, but it may be something that can help you. Best of luck, Frank - Original Message - From: "David Backeberg" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, March 3, 2010 12:34:08 PM Subject: [asterisk-users] asterisk SIP, SIPAddHeader() and Cisco GED-125 Greetings: I'm in the situation where I'm trying to splash information picked off by an asterisk IVR into a Cisco call center environment. I'm under the impression that the ONLY way to do this is to setup socket connections with the Cisco "voice processor", or CVP, and send packets corresponding to GED-125. Cisco has a detailed 100+-page document detailing the internals of what these packets need to look like. But wouldn't it be nice if instead, you could use SIPAddHeader() with X tags and have Cisco pick off the out-of-band values from SIP packets? Wouldn't it be even nicer if there was a middleware that spoke GED-125 out of one side, and spoke SIP X headers on the other side? I will soon be able to tell you about the bowels of this interaction, but before I go down this road, does anybody want to speak up with lessons learned from doing this themselves? I'm assuming I'm going to end up creating a library in Perl to help me do this (that is, the out-of-band conversation with the CVP). -- _ -- 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 -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 fw...@imminc.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] Meetme voice quality problems
ztdummy can give you issues as a timing device. Any way you could try using a
Digium card just as a timing device to see if this helps?
- Original Message -
From: "Tomasz Zieleniewski" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, January 30, 2008 11:23:57 AM (GMT-0500) America/New_York
Subject: [asterisk-users] Meetme voice quality problems
Hi,
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is "cut".
Each voice sequence is disturbed.
Does any one have similar issue and could give me some advice??
my extension.conf for meetme:
;switch => Realtime/macro-conference
exten => s,1,NoOp(-- Macro-conference for:${MARCO_EXTEN} start --)
exten => s,n,Answer
exten => s,n,Wait(1)
exten => s,n,MeetMe(|cdIps)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup
Thank for any help.
Kind Regards
Tomasz
--
Franklin Webb
Asst Project Manager
Inter Medi@ Marketing Solutions
610-701-9670
[EMAIL PROTECTED]
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Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite
Thanks to both of you for your input. I'll be in touch off list Steve. -Franklin - Original Message - From: "Steve Totaro" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, January 30, 2008 8:00:48 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite On Jan 29, 2008 8:36 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > On Jan 29, 2008 5:55 PM, Alex Balashov <[EMAIL PROTECTED]> wrote: > > > > Franklin, > > > > Because ChanSpy() is a "passive" monitor, there is nothing about the > > implementation that would cause Asterisk to shunt the speech back to > > itself. Asterisk only does this in situations where it is out of the > > media path and needs to insinuate itself back into it for the purpose > > of generating media, such as on-hold music, IVR, etc. > > > > What you're wanting should, in my opinion, basically be submitted as a > > feature request. Perhaps the developers can add a flag to the ChanSpy() > > invocation repertoire to make this work. > > > > Cheers, > > > > -- Alex > > > > -- > > Alex Balashov > > Evariste Systems > > Web: http://www.evaristesys.com/ > > Tel: +1-678-954-0670 > > Direct : +1-678-954-0671 > > > Alex, he was not asking why, it is obvious he knows why. > > He was asking for a solution or idea on how to work around this issue. > > Are you using Sangoma cards? If so, I might have a very good answer > for you, as well as another very possible different solution. Both > would be outside of Asterisk so some kind of magic would have to > happen to associate the call being spied on to the channel but that > should not be that difficult if you even need it. > > Another solution is to track down the code referenced here > http://bugs.digium.com/view.php?id=9888 and modify chanspy to do a > reinvite back to asterisk before starting the spy. > > Anyways, I am sure it can be done. The question is how much time is > it worth to make it happen. > > Maybe we should meet for lunch this week. I can meet you in cow > country or Philly if you want, your choice. I have to go to both this > week anyways and would like to catch up with things since Astricon. > > Thanks, > Steve Totaro > I just confirmed that there is a solution that is perfect for this that has been developed with a web interface to select the call to monitor. A little added code and you can pretty easily look up who the agent handling the call is. Let's test it out on your call center. Again, it is not an Asterisk app and would have no impact on your operations if it does not work. Thanks, Steve Totaro ___ -- 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 -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] ___ -- 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] chanspy does not pull the call back to asterisk after a reinvite
Hello all, I am allowing a reinvite between a snom 320 phone and a SIP gateway to take load off my Asterisk server. When I put the caller on hold, for example, Asterisk successfully reinserts itself into the rtp stream to play music on hold to the caller, but when I do a chanspy Asterisk does not seem to pull the call back. If I am spying on a channel when the call build up happens the reinvite never occurs and it works, but I cannot jump in and spy on a call in progress once the reinvite has happened. Has anyone run into this issue any maybe have a solution, or does anyone know of a good way to get that call back onto the Asterisk switch from another extension prior to calling chanspy? Thanks much, Franklin Webb -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] ___ -- 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] SNOM missed call.
Hi Thomas, That setting is controlled by line. Maybe you could setup two seperate lines on the phones and direct the two different call types accordingly. Franklin Webb Assistant IT Project Leader Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] - Original Message - From: Thomas Laurids Pedersen <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 20, 2006 2:19:14 PM GMT-0500 Subject: [asterisk-users] SNOM missed call. Hi All, Using AAH 2.8. I have configured a group to handle a common number for a remote office. All phones in the office is in the group and they are ringing with a seperate ringtone. All this is very well. However all phones other than the one how answered the call is recording a missed call. I know this is an option in the SNOM phone, but is there some way to avoid this for this type of calls ? or is there another way of doing this ? Best regards Thomas Laurids Pedersen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue errors when phones are down, and possible solution
Greetings all, We run a pretty big Asterisk setup including several satelite offices. We use one central switch and all our offices are connected via MPLS network. We have had some issues with our network at times and it seems like it causes some corresponding problems with Asterisk specifically related to "app_queue.so". Granted we do have some serious volume both in terms of the number of agents we have on the phones and the number of calls comming in on our switch. Previously we were using the "rrmemory" method of distributing our calls, but in an effort to see if it helped with our issues we switched to using the "random" method. Since then we've had some bad network outages and Asterisk seems to hold together far better. The types of problems we had were things such as calls not getting routed to agents (across the board) or agents not being able to log in or out of queues (again across the board). If anyone else has had these issues have you found anything that helps? If you have this problem and have not tried it I would definitely suggest switching your call allocation to "random". Thanks and hope this helps some of you, Franklin Webb Assistant IT Project Leader Inter Medi@ Marketing Solutions 610-701-9670 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unwanted conference with snom320 and asterisk1.07bristuffed
Title: Messaggio We use a large number of Snom 320's and we have this same problem even with Call join on Xfer set to off. I had not previously linked it decisively to the Snoms, but it sounds like that is likely our issue. We've had to stay at the 4.5 firmware because otherwise we get additional incomming calls when our reps put someone on hold to make a transfer. If I find any solution I'll be sure to share it with the list. Thanks for sharing your experiences, Frank Webb Assistant Project Leader Inter Media Marketing Solutions - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, May 04, 2006 7:35 AM Subject: RE: [Asterisk-Users] Unwanted conference with snom320 and asterisk1.07bristuffed Under Advanced make sure this is set: Call join on Xfer (2 calls): to off From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tommaso CalosiSent: Thursday, May 04, 2006 4:02 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Unwanted conference with snom320 and asterisk 1.07bristuffed I have 13 Snom 320 with asterisk 1.07 bristuffed. The problem is that sometimes on random basis, when one customer is placed on hold and another call arrives, the customers are put in conference with each other. This look very strange to me, but I've disabled the confernce button on the snom phones to prevent the human errors, but it still occurs. Investigating I've discovered that a similar problem was fixed with the Snom320 Release 5.2 (http://www.snom.com/snom320_release_notes.html ) It says: fixed unwanted conference bug in offhook/enter during ringback with an incoming call BUT my phones are already running 5.2 firmware. Any idea? Am I the only one with this problem? Do you think is the usual buggy-snom firmware problem? Or it might be an Asterisk problem? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 320 HOLD and TRANSFER not detected
Title: Messaggio Hi there Tommaso, You've probably already tried this, but a reset and reboot often fixes certain flakey behaviours on our Snom 320s. We've had to revert back to the 4.5 firmware because of some issues with the 5.0 and later, mostly that additional calls come in when our reps put customers on hold. I have one phone here still at 5.3 firmware and I tested the hold key and that did not happen. Assuming everything is identicle in the config it could be bad phones, but we've seen a very small percentage of those. Overall we've found the Snom320 to be an excellent phone for our purposes. Best of luck, Franklin Webb Assistant Project Leader Inter-Media Marketing Solutions - Original Message - From: Tommaso Calosi To: asterisk-users@lists.digium.com Sent: Thursday, April 27, 2006 2:03 PM Subject: [Asterisk-Users] Snom 320 HOLD and TRANSFER not detected I have a preoblem with my snom 320 phones. I have 5 snom phones installed and all of them have 5.2 firmware. All have same settings in the advanced panel. On 2 phones when I press the hold or transfer key nothing happens and * does not start the musiconhold. In the The hold and transfer keys are set as F_R and F_TRANSFER correctly as the others. Other snoms and gxp-2000 work ok. Any ideas? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues - Dumb question
- Original Message - From: "Wes Baehr" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, April 03, 2006 3:16 PM Subject: [Asterisk-Users] Queues - Dumb question > It was my understanding that when an agent answers a queue call, he will not > be hit with another call until he finishes his current call. > > Currently, my agents get hit with calls from the queue while they are still > on a previous call, so I've resorted to setting their "call-limit" in > sip.conf to 1. But, this prevents them from putting one call on hold and > making another call (although they could use parking). > > Maybe I misunderstood, but I'm asking anyway :) Wes, I don't know that ours is the best solution, but we addressed this by turning call waiting off on our agents SIP phones. We use Snom 320s and there is a "Call Waiting Indicator" under the advanced section. Like you I originally set the call limit to 1, but that is not feasable because our agents have to place outgoing calls and transfer the caller, and we can't have additional calls incomming during this time. On the bright side one thing I noticed is that if the agent never picks up the second call, the caller remains in the queue, though while they are "ringing the second line" they can't get answered by a different agent until Asterisk decides to give up on that agent and try another. I tested this by "being" the caller and confirmed I never heard anything but music on hold even while the second line on the agent's phone was flashing, though I cannot swear you are getting the same behaviour with your phones. Oh and I also had an issue with Snom Firmware higher than 4.5 that caused this same problem, so we've had to stay with the 4.5 firmware. Hope this is of some help, Frank Webb InterMedia Marketing Solutions Assistant Project Leader ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Snom 320, displaying text on the screenfrom *
Sean, I can concur as far as the comments here regarding the sipsak syntax. We use sipsak to update the display on our phones so our agents know if they are logged in or logged out. The sipsak syntax we use with good results is: sipsak -M -O deasktop -B "(your message)" -r 5060 -s sip:(phone exten)@(phone IP) 5060 is the port we use. In the Snom setup I set the port for each phone to 5060 under the "Advanced" section by putting 5060 in "Network identity (port):" It definitely took me a while to get sipsak working properly, but the good news is once you get it working it is reliable. -Frank Webb Inter Media Marketing Solutions - Original Message - From: Sean Kennedy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, March 09, 2006 7:00 PM Subject: Re: [Asterisk-Users] OT: Snom 320, displaying text on the screenfrom * I have that set, but for some reason I get errors when I try sipsak, and nothing comes through to the phone: sipsak -M -B "test" -s sip:[EMAIL PROTECTED]timeout after 500mstimeout after 500ms...Some debugging info: [EMAIL PROTECTED] root]# sipsak -vvv -M -B "test" -s sip:[EMAIL PROTECTED]warning: ignoring -i option when in usrloc modefqdnhostname: 192.168.1.1our Via-Line: Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;aliasNew message with Via-Line:MESSAGE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;aliasTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 1 MESSAGEContent-Type: text/plainMax-Forwards: 70User-Agent: sipsak 0.9.5From: sip:[EMAIL PROTECTED]:34213;tag=7c8bd47fContent-Length: 4testsending message ...request:MESSAGE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;aliasTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 1 MESSAGEContent-Type: text/plainMax-Forwards: 70User-Agent: sipsak 0.9.5From: sip:[EMAIL PROTECTED]:34213;tag=7c8bd47fContent-Length: 4testsend to: UDP:192.168.1.67:5060:ignoring MESSAGE retransmissiontimeout after 500 msSo I am at a bit of a loss. Thanks for your help though, I apprecaite it. :)Colin Anderson wrote: Trick with Sipsak is you have to change the network port to 5060 or sipsakmessages never hit the right port. In the web interface, Advaced > AvancedNetwork > Network identity (port): change that to 5060 and you should begood assuming you can figure out sipsak's nasty syntax. hth. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] playing hold time announcement without queue position announcement
Greetings fellow list members, I have what I think is a relatively simple question, but it did not appear to be addressed on the wiki. I am trying to setup a queue so that it plays an estimated holdtime announcement, but not a queue position announcement. Currently my dialplan does both, and while I know how to take out the estimated holdtime without affecting the queue position announcement, I do not see how to do the oppositte. Does anyone know how to do this? Here is a sample of one of my queues from queues.conf: ;; ;;* Development Test Queue * ;; [10001] announce=beep2 ;* a beep to alert the agent of the call servicelevel=30 ;* target service level (maximum time in queue in seconds) musiconhold=default ;* sets music for this queue strategy=rrmemory ;* sets method of allocating calls to reps timeout=20 ;* how long do we let phone ring before it is a timeout retry=5 ;* how long to wait before trying all members again weight=1 ;* weight against other queues sharing agents wrapuptime=4 ;* how long to wait before freeing up for another call maxlen=0 ;* maximum people in queue (0 is no limit) announce-frequency=60 ;* time between position/estimated hold time announcements announce-holdtime=yes ;* announce estimated hold time (yes|no|once) monitor-format=pcm;* record calls in pcm format monitor-join=yes ;* join recordings joinempty=no ;* callers can join an empty queue leavewhenempty=yes;* remove callers from queue if no agents on I know I can set "announce-holdtime" to "no" and remove the hold time, but I'm unsure how to keep the hold time but remove the queue position. in this section of the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+queues.conf you can see it says: ; How often to announce queue position and/or estimated holdtime to caller (0=off) ; ;announce-frequency = 90 ; ; Should we include estimated hold time in position announcements? ; Either yes, no, or only once; hold time will not be announced if <1 minute ; ;announce-holdtime = yes|no|once the fact that is says "and/or" leads me to believe there is a way to only play the hold time without the queue position, but I do not see any suggestions on how to do this. Thanks in advance for any advice, Franklin Webb Inter Media Marketing Solutions ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: setting outgoing caller ID by the queue an extension is logged into
Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, we log in by the extension), and frequently these agents have to make attended transfer calls to outside numbers. This transfer basically amounts to a new outgoing call. I have been asked to set the caller ID for these outgoing calls based on the queue the phone representative is currently logged in to. Unfortunetly I cannot think of a way to do this. The incomming and outgoing calls are two different calls. I have considered using DBPut and DBGet to store this information in a database. This might work, but I am also concerned about the overhead involved. I cannot think of a way to do this using global variables since I need to store a seperate value for each extension. Has anyone run into an issue like this and come up with a solution? Any thoughts are much appreciated. Thank you, Franklin Webb Assistant IT Project Leader Inter Media Marketing Solutions ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfers using # in asterisk
Greetings fellow list members, I am using ABE and I am attempting to impliment transfers using "#". I am using both "T" and "t" as options in my Dial() command. I am attempting to hit "#" then enter another extension from my dialplan. I have tried this on both ends of the conversation and also tried hitting "#" again after entering the extension and still no luck. One end of the conversation is a SNOM 320, the other is an outside line. The transfer does not happen, I was wondering if anyone had any suggestions for me, perhaps something easily missed. I've looked at the wiki and I do have canreinvite set to no. Any help or ideas are much appreciated. Thank you, Frank Webb Inter Media Marketing Solutions ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] executing a reload under stress in Asterisk
Fellow list members, I was wondering if anyone else out there has had issues with Asterisk after doing a reload with a number of users on. I fixed a minor bug in my dial plan and did a reload, and I seemed to have a corrupt config file afterwards. I am also considering there may have been some kind of an issue with copying the files. I keep a seperate copy on a test server, and I copied over to Asterisk before doing the reload. I was getting errors that were basically indicating parts of my dial plan were missing (error indicated I was sending control to priorities that did not exist but they should have). I stopped Asterisk and rebooted the machine and the errors still occured. I copied the files over from my test machine a second time, restarted Asterisk, and everything worked beautifully (as it had on the test machine during the whole process). I have some concern the reload may have been related to this problem, which is serious because it pretty much took down our ability to take calls. It seems like it could have been an error occuring in the process of copying hte files also. For the record I ftp the files between machines to a seperate directory and then "cp" them into the asterisk config directory. Any feedback especially on doing reloads with high call volume and using queues etc, would be appreciated. Thanks in advance, Frank Webb Inter Media Marketing Solutions ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] displaying a message on the Snom 320 using sipsak
Greetings fellow list members,
It seems like a lot of
people have been having trouble getting indicators working on the Snom
phones, myself included. Recently I was able to get the "desktop"
functionality of sipsak to work on my Snom320, and I thought I would share what
I could with the list. For those not familiar this will replace the
standard display when you are not on a call (normally showing the registered
extension) with a text message of your choosing. Our intent is to update
this when our agents log into, and out of, queues. This will give a visual
indicator for agents and supervisors in our call center as to whether or not the
phone is logged in, which is a large concern for us, and probably any call
center.
For the record I tried this with a Snom360 also and
could not get it working.
1. Setup the phone in Asterisk as
normal
2. Get and install sipsak. It can be
found at http://sipsak.org/ (can be on any
machine on your network, we used a Fedora Core 3 machine for this).
3. In the Snom320 Configuration, under
the "SIP" tab of your extensions line (Line 1 for me) make sure "Support
Broken Registrar" is set to "on"
4. In the Snom320 Configuration,
under "Advanced" make sure "Filter Packets from Registrar" is set to
"off"
5. In the Snom320 Configuration, under
"Advanced" under "Network identity (port):" set it to "5060" (you
might be able to use a different port in here and in the sipsak command,
but this is what worked for me.
6. Reboot the phone (just to be sure the settings
take)
Then use the following sipsak command:
sipsak -vvv -M -O desktop -B "Test Msg" -r 5060 -s
sip:[EMAIL PROTECTED]
where:
"Test Msg" is the message you
want displayed. To turn the message off just set it to empty string
("").
5060 is the port, you could try
another port here if you set your phone to another port under
"Advanced"
6670 is the extension of the
phone
192.168.51.251 is the IP of the
PHONE, not the Asterisk server. It does not appear that you can use the IP
of the Asterisk server.
You can get a list of phones with IPs using the
Asterisk command "sip show peers". Our intent is to build a simple
database matching extension to IP and then execute sipsak commands from a
script, probably in the manager API, when agents log in and out that will
update the phone display accordingly.
I hope this is helpful to some of you.
Franklin Webb
InterMedia Marketing Solutions
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Re: [Asterisk-Users] updating display of a hardphone based on agentslogging in
I talked to Digium about this and they are saying the best thing may be to get the information from the manager API and display it on a PC if I cannot find a way to get the data into the phone. I plan to keep looking into this, I'll share whatever solution I end up with. Thanks for your suggestion I'll be putting some research into that also, it sounds like it might be the more promising way to actually get it on the phone. - Original Message - From: "BJ Weschke" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, August 25, 2005 10:08 AM Subject: Re: [Asterisk-Users] updating display of a hardphone based on agentslogging in I've been thinking about how one would accomplish the same thing. I've got a CTI enabled GUI that tells the agent that they're logged in with the call centers that I've deployed thus far, but it's not quite the same as the agent just being able to look at the phone as well and know that they're logged in or not. You've got the Snom 320's, so maybe the most straight forward thing to do would be to use the Hint application with them to light a status LED when an agent is logged in and have it go dark when the agent is logged out. Are you using AgentCallBack? I wonder if Hint could be used to status the agent channel itself. Hmmm. Will have to check this out a little bit more. :-) On 8/25/05, Franklin Webb <[EMAIL PROTECTED]> wrote: > Greetings all, > We are settng up a fair sized call center on Asterisk, but we are having > some issues with our agents not knowing if they have logged in and logged > out. Prior to beginning our migration to VoIP the agents logged into our > nortel phones and confirmation was displayed on the phone. > > My question is has anyone out there done anything from Asterisk that can > change the display on a VoIP hardphone? We are currently using the Snom 320 > and the Aastra 9133i. Thus far the only ideas we've had have involved > trying to figure out if you can send back something from the caller ID to > change it on the phone, or maybe I could get away with using SMS to send a > message and that might be enough for the agents. > > Any thoughts or suggestions are much appreciated. > > Thanks, > Frank Webb ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] updating display of a hardphone based on agents logging in
Greetings all, We are settng up a fair sized call center on Asterisk, but we are having some issues with our agents not knowing if they have logged in and logged out. Prior to beginning our migration to VoIP the agents logged into our nortel phones and confirmation was displayed on the phone. My question is has anyone out there done anything from Asterisk that can change the display on a VoIP hardphone? We are currently using the Snom 320 and the Aastra 9133i. Thus far the only ideas we've had have involved trying to figure out if you can send back something from the caller ID to change it on the phone, or maybe I could get away with using SMS to send a message and that might be enough for the agents. Any thoughts or suggestions are much appreciated. Thanks, Frank Webb ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not conforming to the RFC?/Aastra phone delay issue
Fellow list members, I have run into an issue where I encounter a delay at the beginning of a phone conversation when I make outgoing calls through Asterisk with an Aastra 9133i hardphone. This is most noticable when I call a voicemail system with the Aasta and then with a land line or other VoIP phone. The first word or two of the voicemail message is generally cut off. According to Aastra's engineering this is because Asterisk does not confromt o the RFC, setting FTP voice stream before getting the ACK. They have not seen this with other call servers besides Asterisk. Has anyone else seen this sort of behaviour or is aware of this? Right now we are in the process of switching over the business edition, and we are wondering if we will see a difference in this problem. Thanks, Franklin Webb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] question about VoIP headsets used by other call centers
Fellow list members, I have a question about the headsets or headset setup others may have found works best with asterisk for softphones, as well as recommended softphone packages. My company is migrating from traditional call centers to VoIP call centers using Asterisk. We will be replacing our existing phones with softphones. At this time we are considering X-lite and Firefly, but I would be interested to hear what other members running call centers prefer. As far as the headsets go, we are considering a couple of options. We currently use a mix of Plantronic and GN Netcom headsets for traditional phones, and have a lot of call stations, so a lot invested in these headsets. We could purchse VoIP adapters for those headsets and are considering USB (requires some upgrades for us but I understand the quality is notably better), or adapters that connect to a standard PC soundcard. Alternatively we may just purchase all new VoIP headsets, at this point leaning towards Plantronics if we go this route. Any input or information on what other call centers have had success with would be greatly appreciated. Thanks, Franklin Webb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
