Re: [asterisk-users] Just bought a Polycom 501 - Ifeellike myGXP-2000 was better...
On Jul 25, 2006, at 12:52 PM, Mike wrote: I didn't want to start a war either. It was simply an opinion that I thought was worth expressing after reading all those GXP-2000 sucks messages in the past. It's still just an opinion, I am certainly not trying to build a consensus. Thanks for all those who helped me get the phone working. Please do give us another update after you have used the phone for a while? ___ --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 360
Welcome to VoIP... Your operator needs to take care about QoS when you are doing a download. Alternatively, there are some more-or-less tricky and buggy tricks to stop downloads when you are talking; this needs to be done on your IAD. See for example http://www.voip-info.org/wiki-QoS. CS From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid BenderSent: Wednesday, July 26, 2006 12:46 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Snom 360 Hello List, I am trying to configure QoS for the SNOM 360. I plugged the phone in to the internet and then had the customers computer plug in to the phone. Whith default settings when I talked on the phone it was great. As soon as I started a big download the phone call became unclear. I tried messing around with some settings but to no avail. Anyone have any advice ? Thanks. Dovid ___ --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] Connecting branch offices through IPsec tunnel --latency effects?
On 26 Jul 2006, at 02:00, Rich Adamson wrote: Dan Austin wrote: Stephen wrote: If I connect two offices through an IPsec tunnel, what is the impact on latency, and does it noticeably affect calls? That would depend a lot on the equipment that services the IPSEC tunnel endpoints. Has anyone out there tried this? What were the effects? I've run small to mid size offices (20 to 60 people) over IPSEC tunnels during periods of internal network failures with good results. That includes offices on the opposite side of the world with one-way latency normally around 100ms, but often up to 160ms. Using commercial IPSEC endpoints, or OpenSWAN on a decent system only adds a couple of ms, if that. I might add that I did a little research for a non-voip project relative to what cisco 28xx routers could sustain in terms of ipsec- vpn throughput. The cisco doc's report 55 mbs sustained throughput. On the flip side, the older cisco routers can't sustain 500 kbs without adding a hardware encryption board to the router. So, you are probably very right with the depends a lot on the equipment. ;) In some cases it seems that VPN might get you less latency! I hear that some 3g carriers prioritize VPN traffic above VOIP traffic. Tim Panton www.mexuar.com ___ --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] odd sound between SIP IAX clients
On 26 Jul 2006, at 03:04, Joseph Love wrote: The issue which occurs is that the audio from the SIP client to the IAX client will spend most of it's time sounded very robotic, and garbled. It is possible, although very difficult to understand someone who is on the SIP phone. I have asterisk 1.2.10 configured with realtime with both IAX and SIP clients. The SIP clients include a Grandstream gxp2000 hard phone, and Counterpath's X-Lite 3 (for windows) softphone. The IAX clients tested include idefisk (both windows mac), JakenIAX, and LoudHush. GSM is the preferred codec of both IAX SIP clients, and is indeed the codec being used in all tests. Audio from the IAX to the SIP client does not experience any issues. SIP to SIP (and presumably, although untested, IAX to IAX) communication does not experience any issues. We also have a T1 card through which many calls have been placed, both from the IAX and SIP phones, without any audio issues occurring, in either case. If it weren't for that there have been multiple clients tested to verify this robotic sound, I would cough it up to it being a incompatability between the particular clients, but this occurs on all SIP-IAX communication that has been tried. I'm running out of options as SIP-IAX intercommunication is kinda expected (and necessary for me), and out of good softphones for the mac, as most of the mac-compatible softphones are IAX2-based. Please let me know what additional information is needed to help me debug this problem. We have had reports like this, and it is looking like the iax jitterbuffer is the culprit. Try adding jitterbuffer=no to the general section of iax.conf and see if that helps. Tim Panton www.mexuar.com ___ --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] G729 License to Bridge calls through VOIP provider?
Marco Mouta wrote: By the way could any one tell me wich is the Bandwith with IP over head for this codec. about 8kb/s? Let's do some calculations on that: g729a 20ms results in 20 bytes RTP payload in each packet, in order to traverse the OSI model there's some headers that need to be added, as RTP gets an RTP header, the RTP packet gets an UDP header and the UDP datagram gets an IP header: So add a 12 byte RTP header, 8 byte UDP header and a 20 Byte IP header This results in: 20 byte RTP payload + 12 byte RTP header + 8 byte UDP header + 20 byte IP header=60 byte on the ip layer. Thats a 40 byte overhead (so 2/3 of the packet is just headers :) and were still only on the IP layer now) So to transmit just 1 RTP packet you are actualy transmitting 60 bytes on the IP layer, so in order to get the real used bandwidth we need to knowhow many packets we are sending and on which medium (DSL/ethernet/slip/smokesignals): 20 ms results in 50 packets/s so: 50 packet/s *60 bytes/packet=3000 bytes/s that's 300*8=24000 bit/s total bandwidth on the IP layer so the overhead is 24000-8000=16000 bit/s. The fun starts if you are going to send this over DSL, let's continue the calculation: 50 packets of 60 byte IP, add the 2 byte PPPoA header for DSL= 62 bytes per packet. However, DSL operates with 53 bytes ATM cells, in which you can fit 48 bytes payload (and a 5 byte header) so in order to transmit the 62 bytes of data you need: 62/48=2 ATM cells. Why 2 cells you say? Because ATM can't utilize the unused part of cells, so to transmit 62 bytes you use the same amount of bandwith (on dsl) as you would use to transmit 96 (48*2) bytes. So 1 RTP packet uses 96 bytes on the DSL line, as you already know we have 50 packets/s so that's 50 packets/s*2 cells=100 Cells/s 100 cells/s * 53 byte = 53000 bytes/s on the DSL line thats 424000 bits/s to transmit a 8 kbit/s stream :) So the total overhead is 424000-8000=416000 bit/s overhead. If you would use G723 with a 10 ms frequency it gets even worse :) G723 on 10 ms produces 8 byte RTP payload per packet, so with headers that's 48 bytes on the IP layer, but now were sending 100 packets/s so: 48*100=4800 bytes/s -- 38400 bit/s on the ip layer On DSL this would result in 50 byte packets (pppoa header) with won't fit in 1 cell, so you would use 2 cells for each IP packet. 100 packets/s * 2 cells = 200 cells/s that's 200 cells * 53 bytes/cell = 10600 bytes/s on the DSL line 10600*8=84800 bit/s to transmit a 6400 bit/s stream -- 78400 bit/s overhead If you would use G723 with 20 ms (16 byte RTP payload) you only have 42400 at the DSL layer, so by adjusting the sample frequency you could cut the overhead in half :) Erik Versaevel ___ --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 announcement issues
I tried sending this to asterisk-bsd a couple of days ago I've been using Asterisk in various versions on FreeBSD for some time now, but I've only just got to messing around with ACD. I find I can't get any in-queue announcements to work, be they either periodic or queue position announcements. I've read all of the recent posts on this, and the closest thing to my problem I can find is a chap on asterisk-users who reported something similar a month ago, but worked around it by not selecting the r queue option. That doesn't work for me (I wasn't using it anyway) - and I've dropped all queue options as a test, still to no avail. Before I get round to posting my configs for critique, is this a BSD port issue? I see stuff around on the net re the BSD port, to the effect that there are some issues with Asterisk applications which are related to timers. What exactly is meant by that please? Is that what I'm suffering from here or is it something entirely different? My environment is FreeBSD 5.3.18 (in production use), Asterisk 1.2.9.1 from Ports, no hardware telephony cards (using a wholesale IAX provider). Calls are being routed in via IAX2 and the agents are on IAX2 hardphones, not that that latter makes any difference methinks. Many thanks in advance for any help that can be offered. Phil - End forwarded message - ___ --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] Fwd: Problem with chan_zap.so
Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.---BeginMessage--- Hi, i have installed asterisk and VICIDIAL call center and it's working fine couple days but when i reboot the computer there isthe problem.this is the asterisk -vvgc output:[chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': FoundJul 25 17:56:19 WARNING[5799]: chan_zap.c:923 zt_open: Unable to specify channel 1: No such device or addressJul 25 17:56:19 ERROR[5799]: chan_zap.c:6883 mkintf: Unable to open channel 1: No such device or addresshere = 0, tmp-channel = 1, channel = 1Jul 25 17:56:19 ERROR[5799]: chan_zap.c:10319 setup_zap: Unable to register channel '1-2'Jul 25 17:56:19 WARNING[5799]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1Jul 25 17:56:19 WARNING[5799]: loader.c:554 load_modules: Loading module chan_zap.so failed!i really don't now what is the problem. I have TDM400P card and i use it only for timing.There is zapata.confcontext=unused signalling=fxo_ks group=1 channel = 1-2 context=unused signalling=fxs_ks group=2 channel = 3-4Please help. Thank you. Regards Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.---End Message--- ___ --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] Strange error
Hi Friends, We are using "Asterisk" in our office and using "XLite" as softphone and "Teliax" service for USA dialing. Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, my softphone is telling that "I am sorry. That is not a valid extension. Please try again. Error No. 2". But, after sometime, its working fine again without doing anything. My intercom is also working fine always. What is this error? Please tell me the solution. Looking forward to your response. ThanksRegards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --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] ACD Queues Agents logout
hmm looks nicer than mine:
exten = *2002,1,System(asterisk -rx \
agent logoff Agent/ ${AGENTBYCALLERID_${CALLERID}})
exten = *2002,2,Playback(agent-loggedoff)
exten = *2002,3,Hangup
thx for your suggestion, i think i will integrate your solution
regard
KAI
Anthony Rodgers schrieb:
Hi Kai,
This is what we do:
[agent-login]
exten = s,1,NoOp(${AgentUser})
exten =
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loginok)
exten = s,5,Hangup
exten = s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
exten = s,104,Wait(1)
exten = s,105,Playback(agent-loggedoff)
exten = s,106,Hangup
A.
On Jul 20, 2006, at 6:26 AM, Kai Ober wrote:
Okay, I think i have missed something:
When i use AgentCallbackLogin*(||*007) the agent is logged in, fine.
But how do i log OUT.
okay there is a timout,
autologoff=time
but how can an agent explicit log off?
regards
Kai
___
--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
___
--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] Extension planning
Hi all ! I am currently planning a PBX asterisk installation in our new office. We will slowly migrate from our old system to the new system, running both systems paralel. My question is now how to plan the extensions: before we used to have only 2 digit extensions : like 10, 70 etc. I guess for more flexibility we should use 4 digits ? As we will also have asterisk servers in different countries? So for one office I could use 1xxx and for the other 2xxx ? Am I on the right track? Nik ___ --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] Extension planning
Watch your extensions don't conflict with local numbers - in Australia 1XXX numbers are valid! PaulH On Wed, 2006-07-26 at 10:32 +0200, Nik Engel wrote: Hi all ! I am currently planning a PBX asterisk installation in our new office. We will slowly migrate from our old system to the new system, running both systems paralel. My question is now how to plan the extensions: before we used to have only 2 digit extensions : like 10, 70 etc. I guess for more flexibility we should use 4 digits ? As we will also have asterisk servers in different countries? So for one office I could use 1xxx and for the other 2xxx ? Am I on the right track? Nik ___ --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] Asterisk with Linksys SPA-3000
I have setup a SPA-3000 to forward all incoming PSTN calls to the asterisk and for asterisk to use the SPA for outbound calls. This works fine, but is there anyway to make the asterisk call the FXS port? So that I can call the phone when needed and use the PSTN for calls if needed. Thanks, Dean. ___ --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] G729 License to Bridge calls through VOIP provider?
Erik, What a great and detailled explanation! Thank you very much! Ps. If you know anything about legal issues asked abouta g729 please post it here:) Best regards, Marco Mouta On 7/26/06, Erik [EMAIL PROTECTED] wrote: Marco Mouta wrote: By the way could any one tell me wich is the Bandwith with IP over head for this codec. about 8kb/s? Let's do some calculations on that: g729a 20ms results in 20 bytes RTP payload in each packet, in order to traverse the OSI model there's some headers that need to be added, as RTP gets an RTP header, the RTP packet gets an UDP header and the UDP datagram gets an IP header: So add a 12 byte RTP header, 8 byte UDP header and a 20 Byte IP header This results in: 20 byte RTP payload + 12 byte RTP header + 8 byte UDP header + 20 byte IP header=60 byte on the ip layer. Thats a 40 byte overhead (so 2/3 of the packet is just headers :) and were still only on the IP layer now) So to transmit just 1 RTP packet you are actualy transmitting 60 bytes on the IP layer, so in order to get the real used bandwidth we need to knowhow many packets we are sending and on which medium (DSL/ethernet/slip/smokesignals): 20 ms results in 50 packets/s so: 50 packet/s *60 bytes/packet=3000 bytes/s that's 300*8=24000 bit/s total bandwidth on the IP layer so the overhead is 24000-8000=16000 bit/s. The fun starts if you are going to send this over DSL, let's continue the calculation: 50 packets of 60 byte IP, add the 2 byte PPPoA header for DSL= 62 bytes per packet. However, DSL operates with 53 bytes ATM cells, in which you can fit 48 bytes payload (and a 5 byte header) so in order to transmit the 62 bytes of data you need: 62/48=2 ATM cells. Why 2 cells you say? Because ATM can't utilize the unused part of cells, so to transmit 62 bytes you use the same amount of bandwith (on dsl) as you would use to transmit 96 (48*2) bytes. So 1 RTP packet uses 96 bytes on the DSL line, as you already know we have 50 packets/s so that's 50 packets/s*2 cells=100 Cells/s 100 cells/s * 53 byte = 53000 bytes/s on the DSL line thats 424000 bits/s to transmit a 8 kbit/s stream :) So the total overhead is 424000-8000=416000 bit/s overhead. If you would use G723 with a 10 ms frequency it gets even worse :) G723 on 10 ms produces 8 byte RTP payload per packet, so with headers that's 48 bytes on the IP layer, but now were sending 100 packets/s so: 48*100=4800 bytes/s -- 38400 bit/s on the ip layer On DSL this would result in 50 byte packets (pppoa header) with won't fit in 1 cell, so you would use 2 cells for each IP packet. 100 packets/s * 2 cells = 200 cells/s that's 200 cells * 53 bytes/cell = 10600 bytes/s on the DSL line 10600*8=84800 bit/s to transmit a 6400 bit/s stream -- 78400 bit/s overhead If you would use G723 with 20 ms (16 byte RTP payload) you only have 42400 at the DSL layer, so by adjusting the sample frequency you could cut the overhead in half :) Erik Versaevel ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] E1 connectivity question
Dear All, I have bought a digium TE205p in order to move our E1 pri from a siemens pbx to an asterisk server platform, I have already gathered the data needed to configure the card but I am troubled by one thing that seems unclear on all the documents I read. The E1 is currently inserted in a modem and from the modem goes out a cable to the siemens pbx so should I take the E1 from that modem or take the E1 directly from the provider, plus is there any special pin assignment. Your Help will be very much appreciated. -- Thx MAG ___ --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: Conference
Pleasehow can I enable the 3way-conference on a sip gateway like (addpac) by dialing an extension example pressing (*) since the gateway do not have this feature ,I want to make it on server level or if you know the concept of how call-conference work . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] FW: Conference
Khaled, This must be the fifth or sixth time you have posted this question. As I'd written in my earlier mail to you, the reason why you're not getting any reponse is maybe no one has what you want. Keep reposting the same question won't get you the answers. It just annoys everyone. Regards Leo. Khaled Chehab wrote: Please how can I enable the 3way-conference on a sip gateway like (addpac) by dialing an extension example pressing (*) since the gateway do not have this feature ,I want to make it on server level or if you know the concept of how call-conference work . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
RE: [asterisk-users] FW: Conference
If what you are asking for is a conference, you can use MeetMe and transfer the participants to that MeetMe extension. I you want it to be triggered by say the * sign then look at the featuremap in features.conf. Using an AGI and redirect can do this for you. Use the wiki @ www.voip-info.org for reference. Try that first if you get stuck post the relevant code here for help. SNIP ___ --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] Ringing timer
Hi! Does a ringing timer exist in asterisk to control ringing duration? If not, is there a way to control ringing duration? Thanks in advance for your help, Michel Message sent using UebiMiau 2.7.8 ___ --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] Asterisk with Linksys SPA-3000
With pen in hand, Dean @ INKnBITs succussfully stormed bulwarks which others armed with sword and excommunication have been repulsed, and said ... I have setup a SPA-3000 to forward all incoming PSTN calls to the asterisk and for asterisk to use the SPA for outbound calls. This works fine, but is there anyway to make the asterisk call the FXS port? So that I can call the phone when needed and use the PSTN for calls if needed. Thanks, Dean. Try this, it may do what you need. Works for me. http://nerdvittles.com/index.php?p=65 JC ___ --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] Ringing timer
If by ringing duration you mean how long a device will ring, then look at options to Dial If you mean how long the ring sounds to the callee look at indications.conf Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zenone Sent: Wednesday, July 26, 2006 5:38 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Ringing timer Hi! Does a ringing timer exist in asterisk to control ringing duration? If not, is there a way to control ringing duration? Thanks in advance for your help, Michel Message sent using UebiMiau 2.7.8 ___ --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
Re: [asterisk-users] Queue announcement issues
On 07/26/06 14:58 Phil Jordan said the following: Before I get round to posting my configs for critique, is this a BSD port issue? I see stuff around on the net re the BSD port, to the no, it isn't a BSD port issue. many people run asterisk from ports with ACDs without any problems. in your situation, you'd probably need to provide more information (CLI verbose output, for starters) before someone can give you a more accurate solution. effect that there are some issues with Asterisk applications which are related to timers. What exactly is meant by that please? Is that what the zaptel-bsd drivers have the ztdummy timer and they're in ports and subversion. look for zaptel-bsd in the wiki. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] E1 connectivity question
Mohamed A. Gombolaty wrote: Dear All, I have bought a digium TE205p in order to move our E1 pri from a siemens pbx to an asterisk server platform, I have already gathered the data needed to configure the card but I am troubled by one thing that seems unclear on all the documents I read. The E1 is currently inserted in a modem and from the modem goes out a cable to the siemens pbx so should I take the E1 from that modem or take the E1 directly from the provider, plus is there any special pin assignment. Your Help will be very much appreciated. -- Thx MAG If you really mean to say modem then what you are doing will not work. Maybe you mean a CSU/DSU? If it is a CSU/DSU or the box that the telco owns, take the cable coming out of it. Plug it into your asterisk box and see if you get a green light. I suspect you will since it is working with your Siemens box. If not, make an E1/T1 crossover cable. Pinout is: 1 - 4 2 - 5 Thanks, Steve Totaro Thanks, Steve ___ --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] Extension planning
On 7/26/06, Paul Hales [EMAIL PROTECTED] wrote: Watch your extensions don't conflict with local numbers - in Australia 1XXX numbers are valid! And similarly emergency services 3-digit numbers, 112, 999, 911 etc. In fact I would avoid numbers that are even similr to this. 1112 could easily be mis-dialled as 112. Steve ___ --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] Fwd: Problem with chan_zap.so
Hi Ismir. It tries to change of slot its TDM400P, I find that it must resolv. Good luckJosué 2006/7/26, ismir saljic [EMAIL PROTECTED]: Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. -- Mensagem encaminhada --From:ismir saljic [EMAIL PROTECTED]To:[EMAIL PROTECTED] Date:Tue, 25 Jul 2006 09:12:03 -0700 (PDT)Subject:Problem with chan_zap.so Hi, i have installed asterisk and VICIDIAL call center and it's working fine couple days but when i reboot the computer there isthe problem.this is the asterisk -vvgc output: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': FoundJul 25 17:56:19 WARNING[5799]: chan_zap.c:923 zt_open: Unable to specify channel 1: No such device or address Jul 25 17:56:19 ERROR[5799]: chan_zap.c:6883 mkintf: Unable to open channel 1: No such device or addresshere = 0, tmp-channel = 1, channel = 1Jul 25 17:56:19 ERROR[5799]: chan_zap.c:10319 setup_zap: Unable to register channel '1-2' Jul 25 17:56:19 WARNING[5799]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1Jul 25 17:56:19 WARNING[5799]: loader.c:554 load_modules: Loading module chan_zap.so failed! i really don't now what is the problem. I have TDM400P card and i use it only for timing.There is zapata.conf context=unused signalling=fxo_ks group=1 channel = 1-2 context=unused signalling=fxs_ks group=2 channel = 3-4 Please help. Thank you. Regards Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ___--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] E1 connectivity question
Dear Steve, Yes I did mean a csu/dsu I will try your suggestion and update the results. Thx MAG Steve Totaro wrote: Mohamed A. Gombolaty wrote: Dear All, I have bought a digium TE205p in order to move our E1 pri from a siemens pbx to an asterisk server platform, I have already gathered the data needed to configure the card but I am troubled by one thing that seems unclear on all the documents I read. The E1 is currently inserted in a modem and from the modem goes out a cable to the siemens pbx so should I take the E1 from that modem or take the E1 directly from the provider, plus is there any special pin assignment. Your Help will be very much appreciated. -- Thx MAG If you really mean to say modem then what you are doing will not work. Maybe you mean a CSU/DSU? If it is a CSU/DSU or the box that the telco owns, take the cable coming out of it. Plug it into your asterisk box and see if you get a green light. I suspect you will since it is working with your Siemens box. If not, make an E1/T1 crossover cable. Pinout is: 1 - 4 2 - 5 Thanks, Steve Totaro Thanks, Steve ___ --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 -- Thx MAG ___ --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] G729 License to Bridge calls through VOIP provider?
Ps. If you know anything about legal issues asked abouta g729 please post it here:) if you are briding g.729, without transcode, and you will NOT stay in mediapath (canreinvite=yes), you don't need g.729 licence -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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] Sangoma Stops Receiving Calls
Alex Robar wrote: Hi all, I have a Sangoma A200 card with hardware echo cancellation. The card has 12 ports (10 of which are active; All FXO). Twice on this particular card I've seen all ports simply stop receiving incoming calls. There is no other indication of this, however. I am able to place outgoing calls just fine, and call other extensions without issue. When someone calls in, the line just rings and rings, with no indication that the card even sees the calls. I'm not even sure where to begin looking into this. Could anyone give me some pointers as to what I might need to be looking for? I'll be giving Sangoma tech support a call, but if anyone has any debugging pointers, they would be much appreciated. Alex, Does it work if you disable the hardware echo cancellation? I had an A20002D that started to fail after a month or too of normal operation - it would answer PSTN calls, but the callers couldn't hear me, although I heard them. Disabling the HWEC cancellation made things work, but the echo was intolerable. My vendor (Telephonyware) replaced the card (after I tried it in another computer, running another kernel, and testing the original server with a spare A20002D, and cleaning the FXO and HWEC module sockets), and the replacement has worked great since then. - Mike ___ --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] ACD Queues Agents logout
Here is what I do... Exten=777,1,AgentCallbackLogin() Yup, thats it! use your agent id and password, and then enter your dialable number. I say dialable number because you can basically dial any phone number. We have agents that call a toll free number and login to their home phones, pretty sweet! This has to be in the right context to allow this though. Jordan Novak Communications Technician ___ --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] Ringing timer
But my question was, is it possible to free the channel if it rings too long? Michel Message sent using UebiMiau 2.7.8 ___ --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] wip-300 question on audio dial out with tdm2402e
I have a wip-300 linksys phone, tdm2402e and a number of uniden phones. When I call wip-300 to any internal phone it sounds just fine. When I call outside on the tdm2402e board it is not clear. the other person does not hear anything odd but I hear drops in the audio. When I call out with a uniden phone I dont hear any drops in audio. The linksys wip-300 is using the same codec g711u as the uniden phone. Why would I be hearing audio drop outs on the wip-300? ___ --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] Ringing timer
Zenone wrote: But my question was, is it possible to free the channel if it rings too long? Yes. show application dial in the Asterisk CLI will show you where the timeout goes on the Dial line. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Zip code, city and area codes
Is there a table available, which tells me if a zip code, city and area code matches? For now I did it with google, type each info in and found out if it matches, but it would be easier if there is a table available. bye Ronald ___ --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] One way screech or tone
Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] sip realtime
On Wednesday 26 July 2006 00:03, marek cervenka wrote: i'm reading a lot docs about asterisk realtime but i cannot understand how works sip realtime static i need NAT/qualify for SIP. this is not possible with dynamic realtime i want - save data to sql - asterisk -rx reload to read config (sip.conf with sip users) from sql it is possible? For sip realtime static you have probably read: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static However, NAT/qualify for SIP(users) is perfectly possible with dynamic realtime: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip And `sip_buddies` table gives extensive opportunities(including nat and qualify). And there is the advantage of using it(realtime), you do not need to reload when a new user comes. (This is valid for the needed extensions and voicemail attributes, as well) Sorry for twisting a bit your question, but basically realtime static means to store a .conf file into a database(in which case you must delete its equivalent from /etc/asterisk); realtime is when you store users, peers and friends into the database, keeping the skeletons of sipiax2.conf files in /etc/astrisk. In that case the users, peers and friends sip or iax2 info is being read on the fly. The appropriate extensions though, must be addressed with switch = Realtime statement from extensions.conf. Since all .conf files exist they have precedence. Also register= can be done only from a .conf file. Hope it helps. Benchev ___ --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] Zip code, city and area codes
Hi, We purchase the database with zip codes, latitude, longitude, are codes and all for our zip lookup AGI. If you need something simple take a look at http://www.census.gov/geo/www/gazetteer/places2k.html Best regards, Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, July 26, 2006 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zip code, city and area codes Is there a table available, which tells me if a zip code, city and area code matches? For now I did it with google, type each info in and found out if it matches, but it would be easier if there is a table available. bye Ronald ___ --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
RE: [asterisk-users] One way screech or tone
So would this be the remote end echo can freaking out or the Polycom on the caller side? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
Re: [asterisk-users] Zip code, city and area codes
Ronald Wiplinger schrieb: Is there a table available, which tells me if a zip code, city and area code matches? I doubt, that such a table does exist. Imho you will have to look for individual tables for each country. For Germany, look at: http://w3logistics.com Roger. ___ --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] Ringing timer
- Message d'origine De: Eric ManxPower Wieling [EMAIL PROTECTED] A: Zenone [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Objet: Re: [asterisk-users] Ringing timer Date: 26/07/06 12:54 Zenone wrote: gt; But my question was, is it possible to free the channel if it rings too gt; long? Yes. quot;show application dialquot; in the Asterisk CLI will show you where the timeout goes on the Dial line. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. Thanks! I already read 'Unless there is a timeout specified, the Dial application will wait indefinitely until one of the called channels answers, the user hangs up, or if all of the called channels are busy or unavailable. Dialplan executing will continue if no requested channels can be called, or if the timeout expires.' But did the channel answer when its status is 'ringing'? I think yes but I'm maybe wrong. If I'm rigth the timeout option can't help me...What about you? Message sent using UebiMiau 2.7.8 ___ --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] One way screech or tone
IP Phone - Asterisk - PSTN. This would be the Echo Canceler on the Asterisk/Zap - PSTN interface. Bill Gibbs wrote: So would this be the remote end echo can freaking out or the Polycom on the caller side? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Zip code, city and area codes
Quoting Ronald Wiplinger [EMAIL PROTECTED]: Is there a table available, which tells me if a zip code, city and area code matches? For now I did it with google, type each info in and found out if it matches, but it would be easier if there is a table available. If you subscribe to the LERG, you can build a table which might fit your needs. If you simply want to find a zip code for a NPA-NXX, you can lookup the switch for that NPA-NXX in table LERG6 then lookup the zipcode for the switch in table LERG (I think). This works good for finding a nearby zipcode to match a callers ANI. If you need something more than that, it will be difficult. A zip code can serve multiple NPA-NXX's and an NPA-NXX can be in multiple zip codes. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --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] One way screech or tone
Ok, in my case it would be my Cisco 3660 since that's what talks to the PRI. It talks sip to my Asterisk box. Thanks! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone IP Phone - Asterisk - PSTN. This would be the Echo Canceler on the Asterisk/Zap - PSTN interface. Bill Gibbs wrote: So would this be the remote end echo can freaking out or the Polycom on the caller side? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
Re: [asterisk-users] One way screech or tone
Then none of this applies. Bill Gibbs wrote: Ok, in my case it would be my Cisco 3660 since that's what talks to the PRI. It talks sip to my Asterisk box. Thanks! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone IP Phone - Asterisk - PSTN. This would be the Echo Canceler on the Asterisk/Zap - PSTN interface. Bill Gibbs wrote: So would this be the remote end echo can freaking out or the Polycom on the caller side? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] CSTA support for asterisk
Hi, Is it possible to have a CSTA support for asterisk... If possible how to configure it sanchal ___ --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] Just bought a Polycom 501 - Ifeellike myGXP-2000was better...
Will do. I'm already starting to like the Polycom 501, but then again I`m not a typical technically incompetent end-user with no desire to learn anything new. I can see a big learning curve for the customers who go from 5 cascading phone lines on PSTN phones to a VoIP PBX. I believe the transition would be much easier on the GXP-2000 (not taking into consideration the provisioning which does look a lot easier to manage on the Polycom) because it acts 90% like a typical phone. And really, the lack of a backlight is a shame on the Polycom. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: July 26, 2006 2:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Just bought a Polycom 501 - Ifeellike myGXP-2000was better... On Jul 25, 2006, at 12:52 PM, Mike wrote: I didn't want to start a war either. It was simply an opinion that I thought was worth expressing after reading all those GXP-2000 sucks messages in the past. It's still just an opinion, I am certainly not trying to build a consensus. Thanks for all those who helped me get the phone working. Please do give us another update after you have used the phone for a while? ___ --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] Message waiting question...
Hi I have the following setup: SPA3000 (at home) -- Asterisk1 server (at home) --- Asterisk2 server (at work). The reason the SPA3000 isn't connected directly to Asterisk server 2 is because the SPA3000 can't register to more than one SIP account at a time, plus it was more fun that way :) Anyhow, Asterisk1 and Asterisk2 are connected using IAX2. What I would like is to have the SPA3000 Message Waiting indicator based on the voicemail message hosted on the Asterisk2 server. Is this possible? Thanks JY ___ --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] Transfers - No ringback or moh
I get round this bug by replacing: exten = X,1,Dial(sip/blah) with: exten = X,1,Answer exten = X,n,Dial(sip/blah) It means the call is in an answered state before it starts ringing but it doesn't seem to cause any major problems. Mike Martin Schrott - Thinking-Systems wrote: Hi all, I cannot exactly reproduce your problems, but I can tell you, what problem we have on this topic: a calles b. b takes the call and can speak to a. b sets up a attendend transfer (via the softkey configured in asterisk) to c and hears ringing. a hears music on hold. b hears ringing if c answeres and b hanges up, everything is fine. now the problem: if b hangs up, before c has answered (during ringing) a will loose the connection and also be hanged up. I think this should not happen! The transfer should automatically be changed to blind and a should get the ringing played back instead of b. Hope, you can understand my problem and may have any ideas or thoughts. Greetings and Thanks, Martin ___ --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] FS: 2 x Asterisk X100M (red) daughterboard cards - brand new.
I just bought a brand new TDM400P but it came with all 4 cards, not realising I only needed 2. I now have FS: 2 x Asterisk X100M (red) daughterboard cards brand new. Email me your best offer and your location for a confirmed delivered price. (Im in New York 10021). Payment must be made by Paypal. I also have 2 x X100P clone boards, worked fine for the last 2 years, email me best offer for these as well. Cheers, Dean ___ --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] 2 * servers, IAX connection between to dial extensions across IAX - not working
Here is my setup, this is just a test lab til I figure out how to do this
Both machines are on a lan, no routers, firewalls etc between
BoxA
192.168.1.192
2XXX extensions
BoxA iax.conf
[boxb-peer]
username=boxa-user
type=peer
trunk=yes
secret=mypassword
host=192.168.1.139
[boxb-user]
type=user
secret=mypassword2
host=192.168.1.139
context=from-internal
BoxA extensions_custom.conf (included in extensions.conf)
[ext-local-custom]
exten = _1XXX,1,Dial(IAX2/boxb-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
exten = _1XXX,2,Congestion
BoxB
192.168.1.139
1XXX extensions
BoxB iax.conf
[boxa-peer]
username=boxb-user
type=peer
trunk=yes
secret=mypassword2
host=192.168.1.192
[boxa-user]
type=user
secret=mypassword
host=192.168.1.192
context=from-internal
BoxB extensions_custom.conf (included in extensions.conf)
[ext-local-custom]
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
exten = _2XXX,2,Congestion
calling the 2xxx extensions gets me the following message no matter what extension I call in that group
In this case the number I was dialing is 2001
-- Executing Dial(SIP/2001-ea9d, IAX2/boxb-user:[EMAIL PROTECTED]/001|30|r) in new stack
-- Called boxb-user:[EMAIL PROTECTED]/001
-- Hungup 'IAX2/boxb-peer-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion(SIP/2001-ea9d, ) in new stack
-- Executing Macro(SIP/2001-ea9d, hangupcall) in new stack
-- Executing ResetCDR(SIP/2001-ea9d, w) in new stack
-- Executing NoCDR(SIP/2001-ea9d, ) in new stack
-- Executing Wait(SIP/2001-ea9d, 5) in new stack
-- Executing Dial(SIP/2001-3bd5, IAX2/boxb-user:[EMAIL PROTECTED]/002|30|r) in new stack
-- Called boxb-user:[EMAIL PROTECTED]/002
-- Hungup 'IAX2/boxb-peer-2'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion(SIP/2001-3bd5, ) in new stack
-- Executing Macro(SIP/2001-3bd5, hangupcall) in new stack
Does it look correct? Am I missing something in this config?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
change ${EXTEN:1} to ${EXTEN}
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working
Your cutting the leading dialed number from each box
exten =
_2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
exten = _2XXX,2,Congestion
should be
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN},30,r)
exten = _2XXX,2,Congestion
Bails
Tim P wrote:
Here is my setup, this is just a test lab til I figure out how to do this
Both machines are on a lan, no routers, firewalls etc between
BoxA
192.168.1.192 http://192.168.1.192
2XXX extensions
BoxA iax.conf
[boxb-peer]
username=boxa-user
type=peer
trunk=yes
secret=mypassword
host=192.168.1.139 http://192.168.1.139
[boxb-user]
type=user
secret=mypassword2
host=192.168.1.139 http://192.168.1.139
context=from-internal
BoxA extensions_custom.conf (included in extensions.conf)
[ext-local-custom]
exten =
_1XXX,1,Dial(IAX2/boxb-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
exten = _1XXX,2,Congestion
BoxB
192.168.1.139 http://192.168.1.139
1XXX extensions
BoxB iax.conf
[boxa-peer]
username=boxb-user
type=peer
trunk=yes
secret=mypassword2
host=192.168.1.192 http://192.168.1.192
[boxa-user]
type=user
secret=mypassword
host=192.168.1.192 http://192.168.1.192
context=from-internal
BoxB extensions_custom.conf (included in extensions.conf)
[ext-local-custom]
exten =
_2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)
exten = _2XXX,2,Congestion
calling the 2xxx extensions gets me the following message no matter what
extension I call in that group
In this case the number I was dialing is 2001
-- Executing Dial(SIP/2001-ea9d,
IAX2/boxb-user:[EMAIL PROTECTED]/001|30|r) in new stack
-- Called boxb-user:[EMAIL PROTECTED]/001
-- Hungup 'IAX2/boxb-peer-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion(SIP/2001-ea9d, ) in new stack
-- Executing Macro(SIP/2001-ea9d, hangupcall) in new stack
-- Executing ResetCDR(SIP/2001-ea9d, w) in new stack
-- Executing NoCDR(SIP/2001-ea9d, ) in new stack
-- Executing Wait(SIP/2001-ea9d, 5) in new stack
-- Executing Dial(SIP/2001-3bd5,
IAX2/boxb-user:[EMAIL PROTECTED]/002|30|r) in new stack
-- Called boxb-user:[EMAIL PROTECTED]/002
-- Hungup 'IAX2/boxb-peer-2'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion(SIP/2001-3bd5, ) in new stack
-- Executing Macro(SIP/2001-3bd5, hangupcall) in new stack
Does it look correct? Am I missing something in this config?
___
--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
Re: [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working
while that did seem to change the error message it still doesn't ring the other phone
here is the error message:
-- Executing Dial(SIP/2001-781d, IAX2/boxb-user:[EMAIL PROTECTED]/1001|30|r) in new stack
-- Called boxb-user:[EMAIL PROTECTED]/1001
-- Hungup 'IAX2/boxb-peer-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion(SIP/2001-781d, ) in new stack
-- Executing Macro(SIP/2001-781d, hangupcall) in new stack
-- Executing ResetCDR(SIP/2001-781d, w) in new stack
-- Executing NoCDR(SIP/2001-781d, ) in new stack
-- Executing Wait(SIP/2001-781d, 5) in new stack
Do I need to add something to tell it to use IAX to get to the other server but that it should be ringing a sip extension?
On 7/26/06, whois wes [EMAIL PROTECTED] wrote:
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/${EXTEN:1},30,r)change ${EXTEN:1} to ${EXTEN}___--Bandwidth and Colocation provided by
Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 * servers, IAX connection between to dial extensions across IAX - not working
IAX2/boxb-user:mypassword2 at 192.168.1.139/1001|30|r) in new stack
it's dialing extension 1001, not 2001
ah, i seetry this (dirty, but it should work)
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/2${EXTEN:1},30,r)
___
--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] 2 * servers, IAX connection between to dial extensions across IAX - not working
Great that fixed it, i can now call to the 2xxx extensions from my 1xxx extensions
awsome, thanks so much!
On 7/26/06, whois wes [EMAIL PROTECTED] wrote:
IAX2/boxb-user:mypassword2 at 192.168.1.139/1001|30|r) in new stackit's dialing extension 1001, not 2001ah, i seetry this (dirty, but it should work)
exten = _2XXX,1,Dial(IAX2/boxa-user:[EMAIL PROTECTED]/2${EXTEN:1},30,r)___--Bandwidth and Colocation provided by Easynews.com
--asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension,
if the extension hangs up first, the next line of the dialplan gets
executed,
if the pstn hangs up first, shows exited non-zero on ZAP/6-1 and
the next line doesn't get executed. ( 3,system(...) )
this is my dialplan
exten =_X.,1,Answer
exten =_X.,2,Dial(Zap/g1/${EXTEN},,tTrhH)
exten =_X.,3,system(/bin/echo -e '${DIALEDTIME}, ${ANSWEREDTIME},
${TIMESTAMP},${CALLERID},${EXTEN},${CHANNEL}' /home/mque/E1/list_calls)
exten =_X.,4,Congestion
___
--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] MWI from Octel to Asterisk
2006/7/26, Mike Diehl [EMAIL PROTECTED]: We have ISDN phones that have a Message Light that we don't want to break.Hi Mike,How will these phones be connected ?Regards ___ --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] SIP is not working sometimes. IAX is working fine. Why?
Hi, We are using "Asterisk" in our office and using "XLite" as softphone and your service for making calls to USA. When I am using SIP, Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, it is telling that "I am sorry. That is not a valid extension. Please try again. Error No. 2". But, after sometime, its working fine again without doing anything or any modifications. My intercom is also working fine always. What is this error? Please tell me the solution. When I am using IAX, It is working fine always. What is the problem with SIP?Looking forward to your response.ThanksRegards, Chandra. See the all-new, redesigned Yahoo.com. Check it out. ___ --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] CSTA support for asterisk
2006/7/26, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hi, Is it possible to have a CSTA support for asterisk... If possible how to configure it sanchalHiFor curiosity, why would you like Asterisk to support CSTA ?Do you have any legacy applications or devices needing it ?Regards ___ --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] Zip code, city and area codes
If you need something more than that, it will be difficult. A zip code can serve multiple NPA-NXX's and an NPA-NXX can be in multiple zip codes. Don't forget that number portability significantly muddied the waters, and VoIP has created an environment where there's no longer any need for a physical relationship. For example, ipKall offers Washington state phone numbers for free. Probably unusual to find your average end user having such a number, but some people on this list will. Number portability probably means that you either need a bit of fuzzy logic to determine if the ZIP and the NPA-NXX are in the same region, or you need to be clever about the implementation and how you handle negative matches. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] Which ATA to test T.38 ? What about Linksys 3102
Hi,Which ATA supporting T.38 would you recommend (for reliability) ?Has anyone experienced this one ?http://www.voip-info.org/wiki/index.php?page=Linksys-Cisco+3102 Regards ___ --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] Rookie voicemail user question
On 7/25/06, William Piper [EMAIL PROTECTED] wrote: On 7/25/06, Randy Paries [EMAIL PROTECTED] wrote: Hello, I just got my Asterisk up and running, and everything is great What i can not seem to find is a doc that describes any of the user commands Like is there things like, end message or listen to the message i am leaving , or anything like that? Thanks Randy Google is your friend, learn to use it: http://www.google.com/search?hl=enq=asterisk+voicemail+menu bp ___ --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 Correct this is for listening to voicemails that have been created My questions was, are there any commands to use while recording the voicemail. ___ --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] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...
On Mon, 24 Jul 2006, Douglas Garstang wrote: Not for our users. We held focus groups, and the Polycom's won in terms of ease-of-use over all the other phones investigated. Which other phones did you investigate specifically? Our users found the polycom menus cumbersome, with commonly used options buried 3 or more levels deep. Transfers don't work the way users expect (blind vs attended), and other issues. -Dan ___ --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] Polycom_acd_functions SIP trouble
James Fromm wrote: Yeah, we tried that. Tried every combination of variables in sip.conf. Only solution that works is removing the requirement for a secret. Faris Raouf wrote: One thing to try is setting type=peer instead of type=friend. I'm a bit dazed and confused at the moment, but if I remember correctly Polycom phones just don't work with type=friend. Of course this doesn't explain why SJPhone won't work either so maybe I'm totally off-track, but it might be worth giving it a try just the same. Don't give up just yet. I spent hours with exactly the same problem (in the mainstream * release) until I sorted it out with the type=friend. How about re-trying but changing the password in both the polycom and sip.conf? Try a 1 digit password. Also is there no way to get some debug output in Asterisk that can give more details? Something that can show the password being sent and the password expected rather than just saying it is wrong (seems like a very useful thing to have if it isn't there already)? Faris. ___ --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] Just bought a Polycom 501 - I feel likemyGXP-2000was better...
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 26, 2006 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Just bought a Polycom 501 - I feel likemyGXP-2000was better... On Mon, 24 Jul 2006, Douglas Garstang wrote: Not for our users. We held focus groups, and the Polycom's won in terms of ease-of-use over all the other phones investigated. Which other phones did you investigate specifically? Our users found the polycom menus cumbersome, with commonly used options buried 3 or more levels deep. Transfers don't work the way users expect (blind vs attended), and other issues. Cisco 7960, Snom, Sipura ___ --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] odd sound between SIP IAX clients
Hi Tim, Thanks for the suggestion. Unfortunately the jitter buffer does not seem to be the culprit. With it on or off, the issue still occurs. I'm still playing with some testing of codecs to see if it's codec- related, as per Rich Adamson's suggestions, and will continue discussion of that idea once I can have a lengthy conversation with different codecs. Preliminary checks with ulaw seem to hint that it might be codec-related, but I need a longer conversation to verify this. Thanks, -Joe On Jul 26, 2006, at 1:09 AM, Tim Panton wrote: On 26 Jul 2006, at 03:04, Joseph Love wrote: The issue which occurs is that the audio from the SIP client to the IAX client will spend most of it's time sounded very robotic, and garbled. It is possible, although very difficult to understand someone who is on the SIP phone. I have asterisk 1.2.10 configured with realtime with both IAX and SIP clients. The SIP clients include a Grandstream gxp2000 hard phone, and Counterpath's X-Lite 3 (for windows) softphone. The IAX clients tested include idefisk (both windows mac), JakenIAX, and LoudHush. GSM is the preferred codec of both IAX SIP clients, and is indeed the codec being used in all tests. Audio from the IAX to the SIP client does not experience any issues. SIP to SIP (and presumably, although untested, IAX to IAX) communication does not experience any issues. We also have a T1 card through which many calls have been placed, both from the IAX and SIP phones, without any audio issues occurring, in either case. If it weren't for that there have been multiple clients tested to verify this robotic sound, I would cough it up to it being a incompatability between the particular clients, but this occurs on all SIP-IAX communication that has been tried. I'm running out of options as SIP-IAX intercommunication is kinda expected (and necessary for me), and out of good softphones for the mac, as most of the mac-compatible softphones are IAX2-based. Please let me know what additional information is needed to help me debug this problem. We have had reports like this, and it is looking like the iax jitterbuffer is the culprit. Try adding jitterbuffer=no to the general section of iax.conf and see if that helps. Tim Panton www.mexuar.com ___ --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] Sony Ericsson F250m, Sipura 3000 and Asterisk
Hi I have been asked if it possible to connect a SE F250M to Asterisk. I have never used one of these devices before but from what I have gathered they need a FXO interface. As the Asterisk box is hosted remotely would it possible to use a Sipura 3000 to provide the FXO interface and successfully use the F250M. If anyone has any pointers on this I would be grateful. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] Extension planning
And similarly emergency services 3-digit numbers, 112, 999, 911 etc. In fact I would avoid numbers that are even similr to this. 1112 could easily be mis-dialled as 112. sure good point thank you, will adopt that to german emergency numbers, nevertheless 4 digit gives me most flexibility ? Nik ___ --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] Sony Ericsson F250m, Sipura 3000 and Asterisk
I didn't test it with a Sipura, but a TDM400. You can check this page for configuration codes for the F251M. http://blog.julianmenendez.es/configuracion-fct-ericsson-f251m (In Spanish). If the SPA-3000 supports detecting polarity reversals, you'll need them. Julian. On 7/26/06, Jon Farmer [EMAIL PROTECTED] wrote: Hi I have been asked if it possible to connect a SE F250M to Asterisk. I have never used one of these devices before but from what I have gathered they need a FXO interface. As the Asterisk box is hosted remotely would it possible to use a Sipura 3000 to provide the FXO interface and successfully use the F250M. If anyone has any pointers on this I would be grateful. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] Sip phone settings according to logged in user
Hi all ! I am planing to set up around 20 SIP Phones which will be purchased in one bunch, I am more or less free of choice. I wonder if anyone knows sip phones which allow configuration upon login. The following scenario: User logs into any phone and the settings of the phone are always the same. Meaning individual key assignement is always the same. Is this possible with asterisk in combination which any phone or do I require special phones. Nik ___ --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] Queue announcement issues
Thank you for replying, Dinesh.
Right, if all the cryptic comment I found just refers to the Zaptel
stuff, that isn't a problem. Thank you for the clarification. To
business:
My queue timeout is 240 seconds. My periodic announcement interval is
30 seconds. Queue position announcements are disabled (and anyway they
don't work either - which may be a clue).
After 90 plus seconds in the verbose log example to follow, the caller
hangs up, having heard MoH for the duration and the agent's phone
having been ringing just fine. Any insight about where to proceed next
would be most welcome. Thanks.
This is my queue statement in extensions.conf:
exten = s,n,Queue(hasbean-sales,Hthn|||240)
This is my queues.conf:
[general]
persistentmembers = yes
[hasbean-sales]
strategy = ringall
maxlen=0
announce-holdtime = no
announce-frequency = 0
announce = hasbean/salescall
periodic-announce = hasbean/que
periodic-announce-frequency = 30
joinempty = strict
leavewhenempty = strict
. and another queue which is identical in all but name.
This is the verbose log from the call:
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Accepting
AUTHENTICATED call from 193.111.200.135:
requested format = ulaw,
requested prefs = (),
actual format = alaw,
host prefs = (alaw|ilbc|gsm|ulaw),
priority = mine
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Executing
Goto(IAX2/193.111.200.135:4569-4, hasbean-incoming|s|1) in new stack
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Goto (hasbean-incoming,s,1)
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Executing
Answer(IAX2/193.111.200.135:4569-4, ) in new stack
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Executing
Set(IAX2/193.111.200.135:4569-4, TIMEOUT(digit)=5) in new stack
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Digit timeout set to 5
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Executing
Set(IAX2/193.111.200.135:4569-4, TIMEOUT(response)=30) in new stack
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Response timeout set to 30
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Executing
BackGround(IAX2/193.111.200.135:4569-4, hasbean/initialmessage) in
new stack
Jul 26 20:05:13 DEBUG[16371] channel.c: Scheduling timer at 160 sample
intervals
Jul 26 20:05:13 VERBOSE[16371] logger.c: -- Playing
'hasbean/initialmessage' (language 'en')
Jul 26 20:05:13 DEBUG[16371] chan_iax2.c: Ooh, voice format changed to 8
Jul 26 20:05:17 DEBUG[16371] channel.c: Scheduling timer at 0 sample intervals
Jul 26 20:05:17 DEBUG[16371] pbx.c: Oooh, got something to jump out
with ('1')!
Jul 26 20:05:22 VERBOSE[16371] logger.c: == CDR updated on
IAX2/193.111.200.135:4569-4
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Executing
Goto(IAX2/193.111.200.135:4569-4, hasbean-sales|s|1) in new stack
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Goto (hasbean-sales,s,1)
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Executing
Set(IAX2/193.111.200.135:4569-4, CALLERID(name)=Sales Queue) in
new stack
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Executing
Queue(IAX2/193.111.200.135:4569-4, hasbean-sales|Hthn|||240) in
new stack
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Started music on hold,
class 'default', on IAX2/193.111.200.135:4569-4
Jul 26 20:05:22 DEBUG[16371] channel.c: Scheduling timer at 160 sample
intervals
Jul 26 20:05:22 DEBUG[16371] channel.c: Avoiding initial deadlock for
'IAX2/phil-5'
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Called IAX2/phil
Jul 26 20:05:22 DEBUG[16371] channel.c: Generator got voice, switching
to phase locked mode
Jul 26 20:05:22 DEBUG[16371] channel.c: Scheduling timer at 0 sample intervals
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Call accepted by
82.11.45.110 (format gsm)
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Format for call is (gsm)
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- IAX2/phil-5 is ringing
Jul 26 20:05:56 DEBUG[16371] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102:
Match Found
Jul 26 20:06:03 DEBUG[16371] chan_sip.c: Auto destroying call
'[EMAIL PROTECTED]'
Jul 26 20:06:56 DEBUG[16371] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102:
Match Found
Jul 26 20:06:57 DEBUG[16371] chan_iax2.c: Immediately destroying 4,
having received hangup
Jul 26 20:06:57 VERBOSE[16371] logger.c: -- Stopped music on hold
on IAX2/193.111.200.135:4569-4
Jul 26 20:06:57 DEBUG[16371] channel.c: Scheduling timer at 0 sample intervals
Jul 26 20:06:57 DEBUG[16371] chan_iax2.c: We're hanging up IAX2/phil-5 now...
Jul 26 20:06:57 VERBOSE[16371] logger.c: -- Hungup 'IAX2/phil-5'
Jul 26 20:06:57 VERBOSE[16371] logger.c: == Spawn extension
(hasbean-sales, s, 2) exited non-zero on 'IAX2/193.111.200.135:4569-4'
Jul 26 20:06:57 DEBUG[16371] pbx.c: Function result is 'Sales Queue
07798614850'
Jul 26 20:06:57 DEBUG[16371] pbx.c: Function result is
Re: [asterisk-users] Queue announcement issues
On Jul 26, 2006, at 8:58 AM, Phil Jordan wrote: I tried sending this to asterisk-bsd a couple of days ago I've been using Asterisk in various versions on FreeBSD for some time now, but I've only just got to messing around with ACD. I find I can't get any in-queue announcements to work, be they either periodic or queue position announcements. I've read all of the recent posts on this, and the closest thing to my problem I can find is a chap on asterisk-users who reported something similar a month ago, but worked around it by not selecting the r queue option. That doesn't work for me (I wasn't using it anyway) - and I've dropped all queue options as a test, still to no avail. That chap is me ;) I tried this setup on linux, openbsd and osx and everywhere I get the same result. Removing the 'r' option from the Queue dialplan fixed the announcements. I did those tests on the debian packages and the svn 1.2 and trunk. Right now we are looking into recording the 'ringing sound' as moh mp3 :( Before I get round to posting my configs for critique, is this a BSD port issue? I see stuff around on the net re the BSD port, to the effect that there are some issues with Asterisk applications which are related to timers. What exactly is meant by that please? Is that what I'm suffering from here or is it something entirely different? My environment is FreeBSD 5.3.18 (in production use), Asterisk 1.2.9.1 from Ports, no hardware telephony cards (using a wholesale IAX provider). Calls are being routed in via IAX2 and the agents are on IAX2 hardphones, not that that latter makes any difference methinks. again no help, but the issue appears on platforms with sangoma E1 interfaces, junghanns quadbri cards and IAX2 only. ztdummy or not makes no difference. Many thanks in advance for any help that can be offered. If you find a way to get it working with the 'r' option, please share Michiel ___ --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] IAX2 trunking problems
verify property of dev/zap; if your asterisk running in non-root mode; change /dev/zap chown into asterisk non-root user. Regards. Jon Schøpzinsky wrote: Hello list We are having some strange problems. When we setup trunking between two of our servers, the connection only uses trunking one way. Ex: Data From callingserver to receivingserver uses trunking Data from receivingserver to callingserver does not use trunking. I discovered this problem by looking at a tcpdump in Ethereal, and I can see that the trunked meta packets only goes one way. The other way uses normal Mini packets with raw a-law data. Heres the configurations, with password, username and server info removed. Callingserver: [gsmgw1] secret=*** username=** host=*** type=peer trunk=yes notransfer=yes disallow=all allow=alaw allow=g726 Receivingserver: [**] secret=*** context=default host=** type=user accountcode= trunk=yes notransfer=yes Both servers have ztdummy module installed and loaded. Regards Jon Schøpzinsky ___ --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] Sony Ericsson F250m, Sipura 3000 and Asterisk
Julian J. M. wrote: I didn't test it with a Sipura, but a TDM400. You can check this page for configuration codes for the F251M. http://blog.julianmenendez.es/configuracion-fct-ericsson-f251m (In Spanish). If the SPA-3000 supports detecting polarity reversals, you'll need them. Thanks for that.. According to page 50 of this document http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf it does support detecting polarity reversals so it looks promising. I would still be interested in hearing from anyone who actually has it working before purchasing the kit. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] Developing VoIP with Asterisk
Hi Group!Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users.We got an small ISP and we have the project to give telephony (for now) to our users between them. Our resources are limited and I have installed * as a hope to give this service to our users. We have a good network (with small problems) but I believed that is possible to give this service. Our HFC network is very well calibrated and works fine. The users have cable modems to connect to the internet and we give private adresses to some users. I'm searching for someone who has the same problem in the past with similar things, to know how solve it and if is possible to give VoIP calls with a server with a public address and the softphones (for the costs) with extensions registered on our * box. I configured * four months ago and between two extensions and works very well and but later I did the same test on this week and unfortunaly the voice goes out with echo. So I have the feeling that maybe there's something wrong with the codecs and wich codecs do I need to give the service. Thanks for any help you can give meCarlos Bernat ___ --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] Developing VoIP with Asterisk
- Original Message - From: Carlos Alberto Bernat Orozco [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wed, 26 Jul 2006 16:49:03 -0300 Subject: [asterisk-users] Developing VoIP with Asterisk Hi Group! Greetings and salutations. Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users. The thing that you should be thinking about is how many simultaneous calls you want up. It's easy to have 500 accounts on a box. We got an small ISP and we have the project to give telephony (for now) to our users between them. Our resources are limited and I have installed * as a hope to give this service to our users. We have a good network (with small problems) but I believed that is possible to give this service. Our HFC network is very well calibrated and works fine. The users have cable modems to connect to the internet and we give private adresses to some users. This shouldn't be too bad. Do some users have public and some private? You may be able to get away with reinviting internally. This way Asterisk would not handle the audio. I'm searching for someone who has the same problem in the past with similar things, to know how solve it and if is possible to give VoIP calls with a server with a public address and the softphones (for the costs) with extensions registered on our * box. I configured * four months ago and between two extensions and works very well and but later I did the same test on this week and unfortunaly the voice goes out with echo. So I have the feeling that maybe there's something wrong with the codecs and wich codecs do I need to give the service. You should try with a hardphone so you can eliminate one of the variables in the equation. I have heard of problems previously where users were using softphones and they were introducing the echo. Switching to hardphones solved it and narrowed down the problem ;) Thanks for any help you can give me You're welcome and hopefully some others can give some insight and maybe information on their own deployments similar to what you wish to do. Carlos Bernat Joshua Colp Digium ___ --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] problems with IAX, extension recognition and Asterisk 1.2.9.1
Hello all, I was having some trouble earlier with Asterisk mis-hearing my extensions (this is when dialing into a DID from PSTN). For instance, if I dialed 1234 it might hear 122334. I was using Asterisk 1.2.7 and SIP routing at the time, and I upgraded to Asterisk 1.2.9.1 and SIP and things seem to have been fixed. However, I recently noticed the problem occurring again, this time with IAX routing. Using SIP routing, everything seems fine. Anyone else noticed this problem with * 1.2.9.1? thanks, Cory -- web: corybantic.us ___ --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] Which ATA to test T.38 ? What about Linksys 3102
- Original Message - From: Olivier [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Wed, 26 Jul 2006 14:18:29 -0300 Subject: [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102 Hi, Which ATA supporting T.38 would you recommend (for reliability) ? Has anyone experienced this one ? http://www.voip-info.org/wiki/index.php?page=Linksys-Cisco+3102 Regards Hello There, I know that during our testing for T.38 capability in trunk Matt (the person who was doing the testing) went through the Grandstream ATAs initially and could not get them to work. Thanks to a generous donation he then moved onto trying with Sipura ATAs instead and they worked great so I would assume that the 3102 would also work nicely. Sipura (should I call them Linksys now?) have done a good job on their SIP stack and appear to have done a good job on their T.38 implementation too. If you do end up giving them a try, definitely report back so others will have some feedback. Have a great day! Joshua Colp Digium ___ --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] SIP is not working sometimes. IAX is working fine.Why?
- Original Message - From: Crazy Boy [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wed, 26 Jul 2006 13:58:39 -0300 Subject: [asterisk-users] SIP is not working sometimes. IAX is working fine. Why? Hi, We are using Asterisk in our office and using XLite as softphone and your service for making calls to USA. When I am using SIP, Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, it is telling that I am sorry. That is not a valid extension. Please try again. Error No. 2. But, after sometime, its working fine again without doing anything or any modifications. My intercom is also working fine always. What is this error? Please tell me the solution. When I am using IAX, It is working fine always. What is the problem with SIP? We need more information in order to give you an answer (if there is one). Do you mean that when you are using SIP to your provider it sometimes fails? As well, console output would be nice so we could see what your Asterisk is doing. Looking forward to your response. ThanksRegards, Chandra. Joshua Colp Digium ___ --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] Message waiting question...
- Original Message - From: Jean-Yves Avenard [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Wed, 26 Jul 2006 11:19:34 -0300 Subject: [asterisk-users] Message waiting question... Hi Hola! I have the following setup: SPA3000 (at home) -- Asterisk1 server (at home) --- Asterisk2 server (at work). The reason the SPA3000 isn't connected directly to Asterisk server 2 is because the SPA3000 can't register to more than one SIP account at a time, plus it was more fun that way :) Fun is good! Anyhow, Asterisk1 and Asterisk2 are connected using IAX2. What I would like is to have the SPA3000 Message Waiting indicator based on the voicemail message hosted on the Asterisk2 server. Is this possible? Anything is possible, it's just to what extreme do you want to go to make it happen. Right now we have no way of transporting arbitrary information (like MWI status) between servers. In the future however I'm hoping we'll have something. For now there's two er I mean three ways off the top of my head you could approach this. 1. Using ODBC storage to store your voicemail in a database and have each server setup against that database. The MWI will just query the database to see if there are messages, and since there will be... MWI will be sent to the phone. 2. Using the ability to execute an outside application that exists right now and using your own method to communicate back to turn on MWI (maybe generating a SIP NOTIFY to poke the phone with?). 3. Share the voicemail directory over something like NFS. Thanks JY Joshua Colp Digium ___ --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] Strange Error when calling
Dear All, I have a strange problem in recieving calls on the pri the zaptel is green and everything seems very well, but when a call comes I can see the call along with the caller ID but then I get this strange message which make the call hungup: error msg: 'zap-in' from '0109687348' does not exist. Rejecting call on channel 0/18, span 1. the PRI is an E1 and I have the following configuration for extensions.conf [zap-in] exten => s,1,Answer exten => s,2,Dial(sip/100) exten => s,3,Hungup as for the zapata.conf it is as follow: [channels] language=en switchtype=euroisdn signalling=pri_cpe context=zap-in group=0 channel=>1-15,17-31 I don't know what the problem is or where to look, I will appreciate it if someone can help me out? Thx MAG -- Thx MAG ___ --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] E1 connectivity question
Dear Steve, The line has worked like charm, but now I am facing a new problem with recieving the call, I have sent another mail with this issue. Thank you very much for your support Thx MAG "Mohamed A. Gombolaty" wrote: Dear Steve, Yes I did mean a csu/dsu I will try your suggestion and update the results. Thx MAG Steve Totaro wrote: > Mohamed A. Gombolaty wrote: > > Dear All, > > > > I have bought a digium TE205p in order to move our E1 pri from a > > siemens pbx to an asterisk server platform, I have already gathered > > the data needed to configure the card but I am troubled by one thing > > that seems unclear on all the documents I read. > > > > The E1 is currently inserted in a modem and from the modem goes out a > > cable to the siemens pbx so should I take the E1 from that modem or > > take the E1 directly from the provider, plus is there any special pin > > assignment. > > > > Your Help will be very much appreciated. > > > > -- > > Thx > > MAG > > > If you really mean to say modem then what you are doing will not work. > Maybe you mean a CSU/DSU? If it is a CSU/DSU or "the box that the telco > owns, take the cable coming out of it. Plug it into your asterisk box > and see if you get a green light. I suspect you will since it is > working with your Siemens box. If not, make an E1/T1 crossover cable. > Pinout is: > 1 -> 4 > 2 -> 5 > > Thanks, > Steve Totaro > > Thanks, > Steve > ___ > --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 -- Thx MAG -- Thx MAG ___ --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] Just bought a Polycom 501 - I feel likemyGXP-2000 was better...
I really like the IP60x phones. Have started using the IP430, so far after 20 or so they are fine. But the IP30x and 50x I refuse to use. The aastra 480i is also good. The 9133i has promise. I do not like the snoms - any. Grandstream are so so Budgetone is not bad for the price, but not enterprise grade. My evals are based on useability, quality, reliability, and management. pros and cons on all, but the 601/430 are my best picks so far. I have tested and used Cisco also, but their price and license and feature models are nuts, at least the last time I really investigated. On Jul 26, 2006, at 12:27 PM, [EMAIL PROTECTED] wrote: On Mon, 24 Jul 2006, Douglas Garstang wrote: Not for our users. We held focus groups, and the Polycom's won in terms of ease-of-use over all the other phones investigated. Which other phones did you investigate specifically? Our users found the polycom menus cumbersome, with commonly used options buried 3 or more levels deep. Transfers don't work the way users expect (blind vs attended), and other issues. -Dan ___ --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
Re: [asterisk-users] Strange Error when calling
- Original Message -
From: Mohamed A. Gombolaty
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Wed, 26 Jul 2006 18:40:07 -0300
Subject: [asterisk-users] Strange Error when
calling
Dear All,
Greetings.
I have a strange problem in recieving calls on the pri the zaptel is
green and everything seems very well, but when a call comes I can see
the call along with the caller ID but then I get this strange message
which make the call hungup:
error msg: 'zap-in' from '0109687348' does not exist. Rejecting call on
channel 0/18, span 1.
Usually on a PRI you will get the number that the person dialed, the DID or DDI
you might call it. In this case Asterisk will send it to an extension with that
number, not the 's' extension. Try adding an extension with the number that
does the same as your s extension to see if this is it. Or even:
exten = _X.,1,Noop(Hey they called ${EXTEN})
exten = _X.,n,Hangup
--
Thx
MAG
Joshua Colp
Digium
___
--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] Strange Error when calling
This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: Dear All, I have a strange problem in recieving calls on the pri the zaptel is green and everything seems very well, but when a call comes I can see the call along with the caller ID but then I get this strange message which make the call hungup: error msg: 'zap-in' from '0109687348' does not exist. Rejecting call on channel 0/18, span 1. the PRI is an E1 and I have the following configuration for extensions.conf [zap-in] exten = s,1,Answer exten = s,2,Dial(sip/100) exten = s,3,Hungup as for the zapata.conf it is as follow: [channels] language=en switchtype=euroisdn signalling=pri_cpe context=zap-in group=0 channel=1-15,17-31 I don't know what the problem is or where to look, I will appreciate it if someone can help me out? Thx MAG -- Thx MAG ___ --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] Polycom 501 - How to set handset Volume
I have a customer who is HOH (Hard of Hearing) and needs the volume on his handset set to the maximum volume level. Currently he has to manually set the volume to the max on every phone call that he makes which is a pain. How do I set the volume to the max and have the phone remember that volume setting for all future calls? Thanks, -Charles ___ --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] Polycom 501 - How to set handset Volume
Need to configure the volume persist parameter in the config file, I do not think it can be set on the phone directly. On Jul 26, 2006, at 5:04 PM, calvis wrote: I have a customer who is HOH (Hard of Hearing) and needs the volume on his handset set to the maximum volume level. Currently he has to manually set the volume to the max on every phone call that he makes which is a pain. How do I set the volume to the max and have the phone remember that volume setting for all future calls? Thanks, -Charles ___ --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
Re: [asterisk-users] Polycom 501 - How to set handset Volume
find this line in your sip.cfg file. 1= remember last setting, 0=return to default volume voice.volume.persist.handset=1 voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/ At 05:04 PM 7/26/2006, you wrote: I have a customer who is HOH (Hard of Hearing) and needs the volume on his handset set to the maximum volume level. Currently he has to manually set the volume to the max on every phone call that he makes which is a pain. How do I set the volume to the max and have the phone remember that volume setting for all future calls? Thanks, -Charles ___ --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] Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.
Manrique Feoli wrote:
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension,
if the extension hangs up first, the next line of the dialplan gets
executed,
if the pstn hangs up first, shows exited non-zero on ZAP/6-1 and
the next line doesn't get executed. ( 3,system(...) )
this is my dialplan
exten =_X.,1,Answer
exten =_X.,2,Dial(Zap/g1/${EXTEN},,tTrhH)
exten =_X.,3,system(/bin/echo -e '${DIALEDTIME}, ${ANSWEREDTIME},
${TIMESTAMP},${CALLERID},${EXTEN},${CHANNEL}'
/home/mque/E1/list_calls)
You can try to put this in the 'h' extension so it gets executed upon
hangup:
exten = h,1,system(/bin/echo -e '${DIALEDTIME}, ${ANSWEREDTIME},
${TIMESTAMP},${CALLERID},${EXTEN},${CHANNEL}' /home/mque/E1/list_calls)
exten =_X.,4,Congestion
___
--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
--
Andres
Technical Support
http://www.telesip.net
___
--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] Ringing timer
Yes, on a Zap FXO channel, when you can hear ringing, the timeout is counting down, even if the remote party hasn't answered yet. Zenone wrote: - Message d'origine De: Eric ManxPower Wieling [EMAIL PROTECTED] A: Zenone [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Objet: Re: [asterisk-users] Ringing timer Date: 26/07/06 12:54 Zenone wrote: gt; But my question was, is it possible to free the channel if it rings too gt; long? Yes. quot;show application dialquot; in the Asterisk CLI will show you where the timeout goes on the Dial line. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. Thanks! I already read 'Unless there is a timeout specified, the Dial application will wait indefinitely until one of the called channels answers, the user hangs up, or if all of the called channels are busy or unavailable. Dialplan executing will continue if no requested channels can be called, or if the timeout expires.' But did the channel answer when its status is 'ringing'? I think yes but I'm maybe wrong. If I'm rigth the timeout option can't help me...What about you? Message sent using UebiMiau 2.7.8 ___ --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 !DSPAM:500,44c76db5240132002735277! -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] Re: Recording/Monitor after xfer
Hi,
Does any one knows some thing about this issue?
I'll appreciate any comments!
Telles
Rodrigo P. Telles wrote:
Hi,
I'd like to know if some one knows how to make Asterisk record a call after
xfer (not bxfer).
I tried some ways but it doesn't work at all.
extensions.conf example:
exten =
177,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 177,2,Monitor(wav,${CALLFILENAME},bm)
exten = 177,3,Dial(SIP/17,30,tT)
exten = 177,4,Hangup
exten =
178,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 178,2,Monitor(wav,${CALLFILENAME},bm)
exten = 178,3,Dial(SIP/17,30,tT)
exten = 178,4,Hangup
exten =
179,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 179,2,Monitor(wav,${CALLFILENAME},bm)
exten = 179,3,Dial(SIP/17,30,tT)
exten = 179,4,Hangup
Ex:
A = 177
B = 178
C = 179
A calls to B (Monitor starts recording conversation between A and B) and then
B press flash
and calls C (Monitor starts recording conversation between B and C and A stay
on moh) and then
B hangup the phone bridging A with C.
The first (A to B) and the second (B to C) recording ends when B hangup the
phone so I'd like
to have recorded the conversation between A and C, is that possible?
Thanks for any help!
Telles
___
--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] Rookie voicemail user question
On 7/26/06, Randy Paries [EMAIL PROTECTED] wrote: On 7/25/06, William Piper [EMAIL PROTECTED] wrote: On 7/25/06, Randy Paries [EMAIL PROTECTED] wrote: Hello, I just got my Asterisk up and running, and everything is great What i can not seem to find is a doc that describes any of the user commands Like is there things like, end message or listen to the message i am leaving , or anything like that? Thanks Randy Google is your friend, learn to use it: http://www.google.com/search?hl=enq=asterisk+voicemail+menu bpCorrect this is for listening to voicemails that have been createdMy questions was, are there any commands to use while recording the voicemail. The first listing on the google search has the answer that you are looking for. To make this easier for you, I copied pasted the answer you are looking for below: After recording a message (incoming message, busy/unavail greeting, or name) 1 - Accept 2 - Review 3 - Re-record 0 - Reach operator(1) (not available when recording greetings/name) bp ___ --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] Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.
thanks for your worthy advise Andres, that in deed does the trick.
I had actually thought about that solution, but then I'll have to
evaluate all calls again at hangup ( h ) to see how to handle their
end,
That in my case wasnt all that nice given I need different types of
finishing funtions to be performed according to what the call went like
and to the type of call. Besides that it made the dial plan less readable.
I thought maybe if there was a way to avoid this exited non-zero on
ZAP/6-1 situation I could handle each finishing right at each
extensions end.
Andres escribió:
Manrique Feoli wrote:
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension, if the
extension hangs up first, the next line of the dialplan gets executed,
if the pstn hangs up first, shows exited non-zero on ZAP/6-1 and
the next line doesn't get executed. ( 3,system(...) )
this is my dialplan
exten =_X.,1,Answer
exten =_X.,2,Dial(Zap/g1/${EXTEN},,tTrhH)
exten =_X.,3,system(/bin/echo -e '${DIALEDTIME}, ${ANSWEREDTIME},
${TIMESTAMP},${CALLERID},${EXTEN},${CHANNEL}'
/home/mque/E1/list_calls)
You can try to put this in the 'h' extension so it gets executed upon
hangup:
exten = h,1,system(/bin/echo -e '${DIALEDTIME}, ${ANSWEREDTIME},
${TIMESTAMP},${CALLERID},${EXTEN},${CHANNEL}'
/home/mque/E1/list_calls)
exten =_X.,4,Congestion
___
--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
--
**
Manrique Feoli
Gerente Investigación y Desarrollo
[EMAIL PROTECTED]
Kínetos Telefonía e Informática.
www.kinetos.com
506-234-7771
**
___
--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] Which ATA to test T.38 ? What about Linksys 3102
When I looked several months ago, the only Sipura that supported T.38 was the SPA-2100. I haven't searched in a while, but I think it is still true. We go directly from a Cisco gateway to the SPA-2100 and it works great. It is the only ATA that we've seen that works right. Joshua Colp wrote: - Original Message - From: Olivier [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Wed, 26 Jul 2006 14:18:29 -0300 Subject: [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102 Hi, Which ATA supporting T.38 would you recommend (for reliability) ? Has anyone experienced this one ? http://www.voip-info.org/wiki/index.php?page=Linksys-Cisco+3102 Regards Hello There, I know that during our testing for T.38 capability in trunk Matt (the person who was doing the testing) went through the Grandstream ATAs initially and could not get them to work. Thanks to a generous donation he then moved onto trying with Sipura ATAs instead and they worked great so I would assume that the 3102 would also work nicely. Sipura (should I call them Linksys now?) have done a good job on their SIP stack and appear to have done a good job on their T.38 implementation too. If you do end up giving them a try, definitely report back so others will have some feedback. Have a great day! Joshua Colp Digium ___ --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 -- Network stuff you didn't know http://www.networkoblivion.com ___ --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] HP DL380 and the TE4xxP cards
Hello, Does anybody have experience with the Quad T1/E1 PRI cards in an HP DL380? Just a yes it works fine or a never again is enough :-) Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --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] HP DL380 and the TE4xxP cards
On Thu, Jul 27, 2006 at 10:06:35AM +1000, Edwin Groothuis wrote: Hello, Does anybody have experience with the Quad T1/E1 PRI cards in an HP DL380? Just a yes it works fine or a never again is enough :-) I've had a couple of Digium cards in a DL360 working fine, no problems at all. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Sangoma Stops Receiving Calls
Sangoma's response so far is that it is a known issue. They have heard of it from a few customers, but find the issue extremely difficult to produce (and have not yet been able to replicate the problem in their labs, even on their long-term tests). The suggestion (for now) is to update to the latest wanpipe drivers (Beta 6), send all the asterisk and system logs from when the error occured, and see if either of us can get anywhere. So long as I know they're working on it, I'm satisfied for now. AlexOn 7/25/06, shadowym [EMAIL PROTECTED] wrote: What country are you in? Please let us know what Sangoma tells you. From: Alex Robar [mailto:[EMAIL PROTECTED] ] Sent: Tuesday, July 25, 2006 2:12 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Sangoma Stops Receiving Calls Hi all,I have a Sangoma A200 card with hardware echo cancellation. The card has 12 ports (10 of which are active; All FXO). Twice on this particular card I've seen all ports simply stop receiving incoming calls. There is no other indication of this, however. I am able to place outgoing calls just fine, and call other extensions without issue. When someone calls in, the line just rings and rings, with no indication that the card even sees the calls. I'm not even sure where to begin looking into this. Could anyone give me some pointers as to what I might need to be looking for? I'll be giving Sangoma tech support a call, but if anyone has any debugging pointers, they would be much appreciated.Thanks,Alex-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[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] Message waiting question...
Might be easier to share the directory over WebDAV. Only need to have one port open on the work firewall (if in place) to allow access and can also run it over SSL.-brandonOn 7/26/06, Joshua Colp [EMAIL PROTECTED] wrote: - Original Message -From: Jean-Yves Avenard[mailto:[EMAIL PROTECTED]]To: Asterisk Users Mailing List -Non-Commercial Discussion [mailto: asterisk-users@lists.digium.com]Sent:Wed, 26 Jul 2006 11:19:34 -0300Subject: [asterisk-users] Message waitingquestion... HiHola! I have the following setup: SPA3000 (at home) -- Asterisk1 server (at home) --- Asterisk2 server (at work). The reason the SPA3000 isn't connected directly to Asterisk server 2 is because the SPA3000 can't register to more than one SIP account at a time, plus it was more fun that way :)Fun is good! Anyhow, Asterisk1 and Asterisk2 are connected using IAX2. What I would like is to have the SPA3000 Message Waiting indicator based on the voicemail message hosted on the Asterisk2 server. Is this possible?Anything is possible, it's just to what extreme do you want to go to make it happen. Right now we have no way of transporting arbitrary information (like MWI status) between servers. In the future however I'm hoping we'll have something. For now there's two er I mean three ways off the top of my head you could approach this. 1. Using ODBC storage to store your voicemail in a database and have each server setup against that database. The MWI will just query the database to see if there are messages, and since there will be... MWI will be sent to the phone. 2. Using the ability to execute an outside application that exists right now and using your own method to communicate back to turn on MWI (maybe generating a SIP NOTIFY to poke the phone with?).3. Share the voicemail directory over something like NFS. Thanks JYJoshua ColpDigium___--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 -- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --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] HP DL380 and the TE4xxP cards
On Thu, 2006-07-27 at 10:06 +1000, Edwin Groothuis wrote: Hello, Does anybody have experience with the Quad T1/E1 PRI cards in an HP DL380? Just a yes it works fine or a never again is enough :-) It works fine with a TE210P card. I did turn off the hyperthreading. Regards, Patrick ___ --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] Message waiting question...
Hi. Thank you so much for answering. I guess I couldn't get a better qualified answer ! On 7/27/06, Joshua Colp [EMAIL PROTECTED] wrote: Anything is possible, it's just to what extreme do you want to go to make it happen. Right now we have no way of transporting arbitrary information (like MWI status) between servers. In the future however I'm hoping we'll have something. For now there's two er I mean three ways off the top of my head you could approach this. Hum... I'm afraid that what I was expecting 1. Using ODBC storage to store your voicemail in a database and have each server setup against that database. The MWI will just query the database to see if there are messages, and since there will be... MWI will be sent to the phone. This may be a disturbing solution, I have over 30 voicemail on server2 and I guess I would have to convert all of them first. This may be the easiest solution if you can set up a database voicemail for one user only... 2. Using the ability to execute an outside application that exists right now and using your own method to communicate back to turn on MWI (maybe generating a SIP NOTIFY to poke the phone with?). That sounds quite complicated... 3. Share the voicemail directory over something like NFS. How often does asterisk check the content of the voicemail directory? the two machines connect over a 512kbit/s link, I'm afraid there could be a bandwidth problem. JY ___ --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] Cisco 7960 Call Waiting Beep
Anyoneaware of a way to turn off the call waiting beep via tftp for cisco 7960's? Disabling this through the call menu doesn't appear to work.This would be for sip firmware Thanks Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY ___ --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] Which ATA to test T.38 ? What about Linksys 3102
Peder @ NetworkOblivion wrote: When I looked several months ago, the only Sipura that supported T.38 was the SPA-2100. I haven't searched in a while, but I think it is still true. We go directly from a Cisco gateway to the SPA-2100 and it works great. It is the only ATA that we've seen that works right. You mean Linksys kit happily complies with Cisco kit? Will wonders never cease. Has anyone tried using the T.38 support in the Myson Century CS6220 based ATAs with Steves code? ___ --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
