[Asterisk-Users] Phones for vitural office business
I am looking for IP phones that are suited to serviced office operation. The business is to answer calls for customers. The incoming lines are E1 and customers are allocated with DID. So the customers' phone can be answered with the customers' designed messages and instruction. This can be done easily with key phone system. It seems to be a problem for IP phone system. I read previous discussions on the pros and cons of key system and ip phone system. However, still need to offer an solution to the operation whereby when calls come in, the operators will be able to identify whose customer's call and answer for that customer accordingly. I have a look at Snom 220 with console. Does the line appearance function solve this problem if DID is assigned to difference line of the ip phone? Any suggestion will be welcome. David Kwok ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to announce the DNID to the called party
How to announce the DNID to the called party who picks up the phone and say the correct greeting? I suppose it has to say to the called party before the call is bridged. So it has to do something before the dial command transfer the call. Any ideas? David Kwok ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How many line appearance can Snom 200 handle?
Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oz ISDN
In Australia, Telstra, the local telco provides isdn modem for isdn connection. The modem has 2 analogue telephone jacks and a serial port for connection to dialup internet. My question is that will it be possible to use Zaptel TDM02B to connect to the analogue jack instead of getting a fritz card to do the telephony. Will there be less feature if doing so? -- David Kwok, CISSP Tel: 612 82315701 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 CISSP, Certified Information System Security Professional smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] OZ ISDN
Kimble Young wrote: If you go the analogue route: * You'll get poor audio compared to ISDN which is crystal. * Each number will act like a seperate line unlike with an ISDN card where you can receive two calls simultaneously on the same line. * You'll lose cool ISDN features like call deflection. * It won't be as reliable (speculation). * It'll probably cost just as much for two analogue cards as a fritz card. On the positive side you won't have to go through a lot of frustration getting the fritz working. In summary using an analogue adaptor on ISDN rather defeats the purpose of ISDN. You are absolutely spot on. I am hesitated by the sheer amount of configuration with the ISDN driver. The actual implementation is actually even a bit of complicated. The ISDN is used primarily for internet connection and voip from an Australian provider. In that case I need to use whatever driver to intiate dail up to the internet. Will it be isdn4l in this case. Now if and when internet is down for whatever reason, asterisk can still perhaps use the capi driver to connect calls. I don't have much experience in isdn at all. I am not sure where such setup can be done simitaneously. It would be nice if someone can point me to the right direction. The Telstra connection already come with Nt1 +11 modem and I have already got pretty good doc to set it up with redhat 9.0. So I don't have to worry about isdn stuff. I wish to go the correct route which is using Fritz card to do this but I am afraid it is not possible. Regards -- David Kwok, CISSP Tel: 612 82315701 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 CISSP, Certified Information System Security Professional smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] DID/T1
I need clarification as to DID in T1 connection. T1 provides 24 channels for voice/data. Do it assign each channel to particular DID. Or you can have unlimited DID to share the 24 channel as an example. ie. Outgoing/incoming traffic is not bound to particular channel. Whatever is available will be used according to the grouping in zapata.conf. -- David Kwok, CISSP Tel: 612 82315701 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 CISSP, Certified Information System Security Professional smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] illegal instruction - on Via board
I have sorted out the problem of compiling the CVS 040608. When launched the server die with illegal instruction error. Although the Makefile in asterisk is changed to PROC=i586. The Makefile in codecs/ilbc still has a line of reference by using uname -m which will come up with i686 architecture. So if you change that line 1 to: CFLAGS+=-Wall -Werror -fPIC -O3 -march=i586 -funroll-loops ^^ -fomit-frame-pointer LIB=libilbc.a So I wonder the line should be -march=$(PROC) or just hardcord to i586 instead. I am not training in programming, please correct me if I am wrong. This should be submited to asterisk so that they can fix the bug too. It will compile and work correctly. -- David Kwok, CISSP Tel: 612 82315701 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 CISSP, Certified Information System Security Professional smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] illegal instruction -via c5
I rolled back asterisk cvs to 5/25 and it still runs and aborted with illegal instruction I am not too familiar with gdb and not too sure how to trace the illegal instruction. Has anyone have a working cvs version for via hardware? David Kwok ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] change cisco ata 186 dial behaviour
I have ata-186 and grandstream connected to asterisk using sip. I have a voip account with ATP, in Australia. In order to ring HK, I need to dial 0011852. Grandstream behaves normally and send the whole series of digits and it connects ok. But ATA-186 somehow only allow only 11 digits. ON the console it was only 0011852. The last 4 digits got truncated. I have tried another trick. This time I prepend 0011 to the 11 digits. Again Grandstream works correctly. But ATA 186 again only sends 0011852. Very strange indeed. On another matter with ATA- 186, I cannot activate line 2 by putting entry in uid1, there is absolute dead. Would it be hardware issue?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] illegal instruction
I have just compiled the latest cvs 040605 and have this illegal instruction error when launched asterisk. It is compiled on Via c5 processor. In the asterisk/Makefile I have set PROC=i586 but it does not help the situation. Any suggestion. Regards David Kwok smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] Grandstream phone from speaker phone back to handset
I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible to switch back to handset mode. -- David Kwok Tel: 612 82315701 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] iax2 trunk - unable to accept trunk packet
I have 1 * box having x100p installed and the other has no zaptel card at all. both of the * box has compiled in ztdummy module and both have been activated by modprobe ztdummy. When using trunk to connect the 2 *box. The one without zaptel card complaint about unable to accept trunked packet: no matching peer. On the one has zaptel card I have tried to remove the ztdummy module and connect the other * box but still have the same error. Any suggestion? -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] iax2 trunk - unable to accept trunk packet
My * box which has the zaptel card is also connect to voip termination using zaptel card. The same problem exist as with the * box which does not have the zaptel card. The tech director of the voip provider told me that it is not a timing issue and more to do with my config. Let's not cynical about the way I raised the question. I know there is no matching peer. I do not using windoz myself and I am not accustomed to pop up gui. But why there is no matching peer, this is not expected. I presume ztdummy will provide the same timing device as zaptel card. But apparently it is not. Or is it really due to missing of zaptel card on the other peer. My iax.conf is as follows: [general] port=5036 bandwidth=low disallow=all; same as bandwidth=high allow=gsm allow=g729 ;disallow=g723.1; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. jitterbuffer=yes dropcount=3 maxjitterbuffer=500 maxexccessbuffer=100 trunkfreq=20; How frequently to send trunk msgs (in ms) register = iax_home:[EMAIL PROTECTED] trunk=yes tos=lowdelay [iax_office] type=friend #auth=md5 context=int-ext disallow=all allow=g729 trunk=yes The other thing may be helpful is what modules should be loaded without zaptel card. My lsmod show me: [EMAIL PROTECTED] asterisk]# lsmod Module Size Used byNot tainted ztdummy 2532 0 (unused) zaptel179424 8 [ztdummy] soundcore 6404 0 (autoclean) autofs 13268 0 (autoclean) (unused) e100 60644 1 keybdev 2944 0 (unused) mousedev5492 0 (unused) hid22148 0 (unused) input 5856 0 [keybdev mousedev hid] usb-uhci 26348 0 [ztdummy] usbcore78784 1 [hid usb-uhci] ext3 70784 2 jbd51892 2 [ext3] have I got the necessary modules to do trunking? -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] cronjob to reboot gs101
I have used curl to reboot the GS101 as follow: curl -c cookies.txt -dP2=xLogin=Logingnkey=0b82 http://192.168.1.xxx/dologin.htm curl -b cookies.txt http://192.168.1.xxx/rs.htm Put these 2 lines in a script and use cron to reboot everyday. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] cron job to reboot GS101
Does any one regularly reboot GS101? It sometimes lost registration with * and needs to be reboot. What is the best way to do it by cron? David Kwok ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk gui client
I have looked at matt's asterisk gui client at sourceforge. I am not a programmer by trade. The documentation there seems to be a bit lacking. Has anyone have the experience in installing the gui client and may perhaps have a how-to document available for sharing. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] RE: message lights and stutter tones
[EMAIL PROTECTED] wrote: --__--__-- Message: 2 Date: Mon, 8 Mar 2004 20:14:00 - (GMT) From: Simon Chappell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] message lights and stutter tones Reply-To: [EMAIL PROTECTED] Hi al I have 3 GS 101's plugged into asterisk. They work great and teh quality of sound I can not fault. Most people I am speaking to now ask if I have a new phone because the quality is so much better. My latest quandry is to do with the message button and stuttertones. I dont get either.. If i have a message waiting the only way we know is by email.. you have to check what context you set up to voicemail. have a look at your voicemail.conf. In mine: [int-ext] 1001 = 1001,xxx,[EMAIL PROTECTED] then you need to add the context in you sip.conf [1001] maibox = [EMAIL PROTECTED] -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 trunk - no matching peers
I have cvsed to the 3/9/04 version. I have not been able to do trunking with other iax2 server. I was able to do it before. What are the procedures to diagnose this problem? Would it be firewall related? Would it depend on both peers with the same cvs version? -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] flash button on GS101
Has anyone using the flash button on GS101 to access call waiting? My experience is that it does not work. I read in the list that it may need to tweak the flash duration to under 100msec. Has anyone have any solution? -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
*CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Hanging GS101 in a upright position
Has anyone tried to hang GS101 phones on a wall? It has recess holes at the back of the base where you can hang it on a wall. What it lacks is that the handset is not supported for this upright position. Has anyone done any modification on it? I was thinking about velco the handset. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2959 - 10 msgs
[EMAIL PROTECTED] wrote: Message: 10 Date: Mon, 01 Mar 2004 10:02:54 -0500 From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CVS login Reply-To: [EMAIL PROTECTED] Glenn Dalgliesh wrote: I seem to be having trouble with cvs login. anyone having similar problems It just hangs after entering the password Make sure you actually have connectivity to the CVS server (ping/traceroute). Yes it seems to be rather unreliable. Although from FWD to iaxtel seems to be ok. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
Re: [Asterisk-Users] exit
Just use control-c, you will be able to exist and leaving asterisk continue to run in the background. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] RE: Message waiting light not coming on
I cannot get MWI working either with GS101 firmwire 1.0.4.39 My sip.conf has the mailbox number specified. voicemail.conf has mailbox set up. I have collecting mail fine. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Failed to start asterisk
I am using mini-itx motherboard and I installed asterisk stable from
cvs. However below is the messages when starting asterisk by
safe_asterisk. Anyone spotted the cause of not starting.
Last login: Fri Feb 27 10:40:44 2004
[EMAIL PROTECTED] root]# safe_asterisk
[EMAIL PROTECTED] root]# /usr/sbin/safe_asterisk: line 77: 3448 Illegal
instruction (core dumped) asterisk ${ASTARGS} 1/dev/${TTY}
/dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3463 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3478 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3493 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 44: /dev/tty9: Input/output error
/usr/sbin/safe_asterisk: line 45: /dev/tty9: Input/output error
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
smime.p7s
Description: S/MIME Cryptographic Signature
[Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?
|I cannot call out with my SIP phone though. It'll dial, ring my cell |phone twice and then give up and complain that its busy. Even if I try |to answer the cell phone during the first ring. | |Does anyone have a config they could share with me on how to make this |setup work? This sounds like it should be fairly trivial, but I've |beaten my head against the wall on this for a few days. =) | |Thanks alot, |Jason Again most possibily it is codec issue, what sip phone you use and show us your sip.conf. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] codec translation
The route of my call is: gs101--asterisk--iaxtel--asterisk--gs101 I have 2 g729 from Digium and calls to iaxtel can only be in gsm format. The GS101 phones are set to use g729, then 711ulaw. However when the called GS phone is picked up the connection is terminated. These are the console messages: -- SIP/1003-f8e1 is ringing -- SIP/1003-f8e1 answered [EMAIL PROTECTED]:4569]/6 Error Opening channel:2 not available, see va_g729_init_global(..)Feb 24 08:47:55 WARNING[1242768320]: codec_g729b.c:179 lintog729_new: No available g729 resources for channel 2 Feb 24 08:47:55 WARNING[1242768320]: translate.c:111 ast_translator_build_path: Failed to build translator step from 6 to 8 Feb 24 08:47:55 WARNING[1242768320]: chan_sip.c:1322 sip_write: Asked to transmit frame type 2, while native formats is 256 (read/write = 2/256) == Spawn extension (macro-stdexten, s, 4) exited non-zero on '[EMAIL PROTECTED]:4569]/6' in macro 'stdexten' == Spawn extension (incoming, 1003, 1) exited non-zero on '[EMAIL PROTECTED]:4569]/6' -- Executing Hangup([EMAIL PROTECTED]:4569]/6, ) in new stack == Spawn extension (incoming, h, 1) exited non-zero on '[EMAIL PROTECTED]:4569]/6' -- Hungup '[EMAIL PROTECTED]:4569]/6' -- Hungup 'IAX2[69.73.19.178:4569]/5' == Spawn extension (local, 17001813482, 1) exited non-zero on 'SIP/1002-4360' -- Executing Hangup(SIP/1002-4360, ) in new stack My questions are: Although both gs101 are set to use g729 is the actual communication from gs to asterisk using g729 and asterisk to iaxtel using gsm and asterisk to the called gs using g729. Do anyone make sense out of the console messages since I have 2 g729 licence. It should be able to handle 2 g729 channel one receive and one send. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] more codec negotiation problems
I am playing with MeetMe conference. I have 4 Gs101 phones and it does g729A, 711 etc. I have purchase 2 g729 licence. All gs phones are set to use g729. In this case, the system only allow 2 g729 channel for conference. Then I change the GS phone setting to use 711ulaw and still have the same messages: Error Opening channel:2 not available, see va_g729_init_global(..)Feb 24 12:37:38 WARNING[1259545280]: codec_g729b.c:179 lintog729_new: No available g729 resources for channel 2 Feb 24 12:37:38 WARNING[1259545280]: translate.c:111 ast_translator_build_path: Failed to build translator step from 6 to 8 Error Opening channel:2 not available, see see va_g729_init_global(..)Feb 24 12:37:38 WARNING[1259545280]: codec_g729b.c:102 g729tolin_new: No available g729b resources for channel 2 Feb 24 12:37:38 WARNING[1259545280]: translate.c:111 ast_translator_build_path: Failed to build translator step from 8 to 2 Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:462 conf_run: Huh? Got a non-ulaw (256) frame in the conference Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:462 conf_run: Huh? Got a non-ulaw (256) frame in the conference Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:462 conf_run: Huh? Got a non-ulaw (256) frame in the conference Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:462 conf_run: Huh? Got a non-ulaw (256) frame in the conference Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:462 conf_run: Huh? Got a non-ulaw (256) frame in the conference Feb 24 12:37:38 WARNING[1259545280]: chan_sip.c:1322 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:477 conf_run: Unable to write frame to channel: Resource temporarily unavailable Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:477 conf_run: Unable to write frame to channel: Resource temporarily unavailable Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:477 conf_run: Unable to write frame to channel: Resource temporarily unavailable Feb 24 12:37:38 WARNING[1259545280]: app_meetme.c:477 conf_run: Unable to write frame to channel: Re I did a sip show channels at the console and sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 (None) 62f3814d24b 00102/0 0ms ms UNKN 192.168.2.33 1005C9DFEAA8-2E 00101/48922 0ms ms GSM 192.168.1.10410044ffece19584 00101/00067 0ms ms G729A 192.168.1.10110022b02b69070f 00101/51565 0ms ms G729A 4 active SIP channel(s) To my surprise gs are still using g729 to connect with *. And therefore I can only have 2 gs101 phones in conference due to 2 g729 licence. Why aren't gs101 now connect to * using 711u as what I intended?? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] system speed dial
have someone implemented some system memory for speed dial?
due to the lack of speed dial/memory of GS, I feel the need to introduce
some system speed dial function to supplement Gs' deficiency.
I am thinking along:
recording speed dial
exten = _*51.,1,Playback(speed-dial)
exten = _*51.,2,Playback(digits/1)
exten = _*51.,3,DBput(SD/1=${EXTEN:3})
exten = _*51.,4,DBget(temp=SD/1)
exten = _*51.,3,Saydigits(${temp})
exten = _*52.,1,Playback(speed-dial)
.
.
exten = _*61,1,Playback(speed-dial)
exten = _*61,2,Playback(digits/1)
exten = _*61,3,Playback(pls-wait-connect-call)
exten = _*61,4,DBget(temp=SD/1)
exten = _*61,5,Playback(${temp})
exten = _*61,6,Dial(zap/1/${temp},20,tr)
Welcome any suggestions to the above scheme.
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
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[Asterisk-Users] RE: multicasting conference calls
Use the outgoing call feature of asterisk to have the servers join each others conferences. It's very simple. Sorry, I am not quite sure what is the outgoing call feature. Would you please elaborate a bit. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hyuandi pstn handsets
I have 5 Hyuandi pstn handsets and wondering it is possible to plug into tdm400p or ata-286? Perhaps a general question whether handsets from PSTN pbx can be reused with Asterisk? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: codec negotiation prob solved
(Philipp von Klitzing) wrote: FYI - bug 1043 has been fixed on Feb 18: From my log, below, you will see that ast_rtp_bridge is not comparing the codecs properly. Asterisk is currently comparing the integers, and not the bits of the codec. In the below example codec0 = 260. That means Codec0 allows both 256 (g729) and 4 (uLaw). So that would mean that Codec0 and Codec1 do have a Codec Match. Asterisk needs to do a bit compare, and not a int compare in this case. -- SIP/dialnet-8bac answered SIP/chris0-df00 -- Attempting native bridge of SIP/chris0-df00 and SIP/dialnet-8bac Feb 16 11:27:48 WARNING[68889520]: rtp.c:1204 ast_rtp_bridge: codec0 = 260 is not codec1 = 256, cannot native bridge. Feb 16 11:27:50 NOTICE[68889520]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Feb 16 11:27:50 NOTICE[68889520]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 I have the same problem with codec negotiation, my Voip provider use g729 however I have also connection with Iaxtel which only use GSM. I can only get one or the other codec working when dialing out. My iax.conf setting is below: ; Inter-Asterisk eXchange driver definition [general] port=4569 bandwidth=low disallow=all allow=gsm allow=g729 disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. jitterbuffer=yes dropcount=3 maxjitterbuffer=250 maxexccessbuffer=50 register = dkwok:[EMAIL PROTECTED] tos=lowdelay [iax_home] type=friend context=int-ext auth=md5 user=iax_home secret=cc trunking=yes disallow=all allow=gsm host=dynamic qualify=yes [iaxtel] type=friend disallow=all disallow=g729 allow=gsm trunking=yes context=from-iaxtel [atp] type=friend disallow=all allow=g729 trunking=yes context=atp host=xxx.xxx.xxx.xxx I would like to hear any comment from * developer. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] codec negotiation
I have outgoing connection to iaxtel and another iax server A. iax server A only accept g729 codec while iaxtel is something I am not quite sure of. At the moment iaxtel only accepts gsm. I remember previously it does accept g729. my problem due to the switching between codec when making outgoing calls to these servers. my iax.conf has these lines: [general] disallow=all allow=gsm allow=g729 I believe the general context define the codec to be used when making outgoing calls. The peer context below general context is to governed codec to be used for incoming calls. Is this correct? now if I specificly disallow g729 in the general context I can make calls to iaxtel. however i cannot make calls to server A as it only accepts g729. After I allow g729, I can make call to server A but the call made to iaxtel cannot go through. The console indicates that the call is accepted by iaxtel using codec 729A, then it says the circuit is too busy. Is there a clever way of governing the codec use for each outgoing connection in order to avoid the issue in codec negotiation? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: X100P / Echo / ZTMONITOR CAN2,3, etc.
[EMAIL PROTECTED] wrote: The strange thing here is that asterisk is not removing the echo. I did notice that the 0.7.1 tar did not do the echo cancel very well and Mark suggested that I go back to the CVS, which did wonders. You might also verify that echo cancellation is actually turned on. Enter zap show channel x at the CLI, where x is one of your defined zap channels. If it's enabled, somewhere in the output you should see Echo Cancellation: xx taps, Currently On/Off. The On/Off will change from Off to On when a call is bridged. If it is not enabled, check the definitions in zapata and in your make file. I have checked my zapata.conf file and fiddled with the settings so far I cannot get echocancel to work. Although I don't have echo problem now. [channels] ; language=en context=incoming amaflags=default cancallforward = yes callwaiting = no busydetect = no callprogress = no musiconhold = default rxgain=0.0 txgain=0.0 immediate=no echocancel = yes echocancelwhenbridged = yes echotraining = yes usecallerid = yes threewaycalling = yes callwaitingcallerid = yes callerid = asreceived signalling = fxs_ks callreturn = yes channel = 1 At the console, the output for show channel is : Channel: 1 File Descriptor: 19 Span: 1 Extension: Context: incoming Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No I have a look at the Makefile in zapata directory and do not see anything needs to be tweak. David Kwok ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can you savage a failed call transfer
I have a couple of cummsy user who always lose a call when the transfer is not done properly ie due to dialing a wrong digit, etc. My question is that is it possible to savage a failed call transfer? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] x100p dropping incoming calls
I have been experiencing hung up when answering incoming calls through x100p. NOTICE[1242768320]: chan_zap.c:4584 ss_thread: Got event 2 (ring/Answered).. -- Executing Wait(Zap/1-1,1) in new stack -- Executing Answer(Zap/1-1,) in new stack -- Executing DigitTimeout(Zap/1-1.5) in new stack -- Set digit timeout to 5 -- Executing ResponseTimeout(Zap/1-1,5) in new stack -- Set Response Timeout to 5 -- Executing Playback(Zap/1-1,cpswelcom) in new stack -- Playing 'cpswelcome' (language 'en') -- Executing BackGround(Zap/1-1,cpswelcome5) in new stack -- Playing 'cpswelcome5' (language 'en') . . . . -- Called 1001 -- Sip/1001-a924 is ringing -- SIP/1001-a924 answered Zap/1-1 == Spawn extension (operator,s,2) exited non-zero on 'Zap/1-1' -- Executing Hangup(Zap/1-1,) in new stack == Spawn extension (operator, h,1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' As soon I pick up the phone when it rings. The call was hungup. It seems to be occasional and not happens all the time. Anyone has any comment on what possible cause. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] FWD/Iaxtel/Asterisk codec use
The codec issues with different services and sip phone are the most complicated and trusting experience when using Voip services. I had been able to connect to FWD behind a firewall by using Iaxtel using g729. Just recently, about a week, every time I tried to call FWD, the connection simply timed out. The console message says the circuit is busy. Or every one is busy at the moment. However, when I change the codec of the connect to GSM. The connection is back to normal. It may be due to changes to Iaxtel or their G729 licence runs out of capacity. When I fiddled the iax.conf file, in the [iaxtel] section, I specified disallow=all allow=gsm disallow=g729 it still does not work. then I have it change the [general] section as well disallow=all allow=gsm disallow=g729 It then works. Is there any logical explanation to this. I wonder. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Calling from Iaxtel to FWD users always busy
Just recently my calls through iaxtel to FWD user do not go throuhg due to busy circuit. Wonder if there is any change to the setup of Iaxtel? -- Executing Dial(SIP/1002-246A, iax2/xxs:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called xxx:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 69.73.19.178 (format G729A) -- Format for call is G729A -- IAX[69.73.19.178:4569]/2 is circuit-busy -- Hungup 'IAX2[69.73.19.178:4569]/2' ==eVERYONE IS BUSY AT THIS TIME -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2749 - 7 msgs
You can have a look at wiki on iax trunking plus notes on setting up x100p card. David Kwok Message: 5 From: Maninder Bhatia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 5 Feb 2004 16:12:07 -0500 Subject: [Asterisk-Users] X100P - Asterisk - Asterisk - X100P setup help Reply-To: [EMAIL PROTECTED] I am very new to this forum, and to Asterisk world. I have two two X100P cards, and was trying to=20 setup something which looks like=20 phone line (PSTN) -- Asterisk X100P card -Asterisk (Linux Box 1) = -- =20 Asterisk X100P card -Asterisk (Linux Box 2) -- phone line (PSTN) Was requesting if someone could confirm if this could be done, and if = it can be can I get the config file to do this, or the direction I should be going = on.=20 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] RE: setting up ---- newbie
Date: Sun, 1 Feb 2004 18:24:51 -0400 Subject: [Asterisk-Users] setting up newbie Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_00A4_01C3E8F0.AED1DC90 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable |questions : |how can i do this ? what are the commands for this simple setup ? |how can i place calls using a webbrowser (explorer, etc ?) |can i use messenger to call to call my pstn port ? |can i translate h323 to sip and viceversa ? As a newbie I will introduce you to go to www.voip.org web site to read some of the newbie articles and also the documentation on www.digium.com. There is a lot info which will definitely answer your above questions. When you are stuck with your work please do not hesitate to answer specific questions. No doubt everyone here on the list will lend a helping hand. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Transfer of call from a call queue
I have a call queue set up. The phones are xlite soft phone and Gs101 hard phone. I have AgentCallback setup. After the call is answered by an agent, it cannot transferred to another extension. As soon as transfer is executed, the call is hung up. Is this normal behaviour or I miss something. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Call Queues
I have setup AgentCallbackLogin and the agents have been logged in successfully. However when calls are queued and an agent picks up the call. It just hang up the call. On the command console it does say the agent agent 1001 hang up on customers. they must be pissed off. I agreed. My queues.conf file: [agents] ackcall=no agent = 1001,1001,xx ss My queues.conf file: [incoming] announce = incoming strategy=ringall musice = default member = Agent/1001 member = Agent/1002 My extensions.conf : exten = 28,1,AgentCallbackLogin(|@local) exten = 29,1,Queue(incoming) In order to annonce to agent the correct queue does it have to have a gsm file to playback the name of the queue ie incoming in this case? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Call Queues
I have setup AgentCallbackLogin and the agents have been logged in successfully. However when calls are queued and an agent picks up the call. It just hang up the call. On the command console it does say the agent agent 1001 hang up on customers. they must be pissed off. I agreed. My queues.conf file: [agents] ackcall=no agent = 1001,1001,xx ss My queues.conf file: [incoming] announce = incoming strategy=ringall musice = default member = Agent/1001 member = Agent/1002 My extensions.conf : exten = 28,1,AgentCallbackLogin(|@local) exten = 29,1,Queue(incoming) In order to annonce to agent the correct queue does it have to have a gsm file to playback the name of the queue ie incoming in this case? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Grandstream 100 sidetone
For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the volume of the voice call. As the sidetone is louder than the conversation it is getting rather distracting. Can the sidetone be calibrated or adjusted? If not, how are people coupling with it? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
Subject: Re: [Asterisk-Users] Grandstream 100 sidetone
Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Grandstream 101
Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060 dtmf = sip info codec = pcmu codec = pcma Any pointer of a sample of config file would be most appreciate. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Grandstream 10
Resolved my problem: somehow, the setting of sip.conf has to be in this order [general] disallow=all allow=alaw allow=ulaw allow=gsm The codec in use is alaw. It works but I would like to use gsm if possible. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] sidetone issue
I am using GS 101 and as I am new to Ip phone arena. I am finding it a bit annoying to hear sidetone, especially when both parties are talking over each other occassionally. In that case, I cannot hear the other party's conversation. Is there any way to suppress it? Is it only GS or it applies to more expensive phone eg Cisco 7960 as well? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Brandwidth for making internet calls
My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Zap show channel
What are the meaning of these Zap show channel output? Caller ID string: Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Are these settings configurable in /etc/asterisk/zapata.conf? -- David Kwok Iaxtel/FWD # 17001813482 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] echo cancellation
echo cancellation is activated in /etc/asterisk/zapata.conf However, how to confirm it? Does zap show channel 1 confirm the existence of echo cancellation? -- David Kwok Iaxtel/FWD # 17001813482 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] words for Alison
Call forwarding Call forwarding not reply call forwarding busy David Kwok smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] hardware requirements - asterisk
In relation to voice degradation when having 2 or more connection to Asterisk. The comment on the network setup is quite possible. I am not too familiar with linux. How do I check whether the asterisk server's nic is running at full-duplex mode. Does Asterisk use the sound card on the box to do voice processing? I am running xlite on 2 pc and making calls through iax, FWD and back to my incoming call menu. Voice degradation happens. David Kwok smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] hardware requirements of asterisk
I have been playing with 2 Asterisk boxes for testing purposes, it has been going very well. The 2 boxes are PII celeron 400 (HP Deskpro) with sound cards and lan. I have iax connecting the 2 boxes. For making cals and testing out recorded message for 1 connection it was working quite well. However, when I stressed it a bit with 2 users making calls, we started to here voice degradation and cracking noises. However, top shows cpu is 94% idle. I am suspecting the network. However it is 100M switch and I have not had any clue. I suppose it should at least be able to handle 10 calls similtaneously for even a small office. So what is the recommended spec for 5 users or 10 users? David Kwok smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] cisco 7910 phone
Hi All Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2413 - 13 msgs
--__--__-- Message: 1 From: Terence Parker [EMAIL PROTECTED] Date: Fri, 9 Jan 2004 11:25:23 +0800 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem registering FWD Reply-To: [EMAIL PROTECTED] --Apple-Mail-1-822243116 Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset=US-ASCII; format=flowed I seem to have a problem registering my Asterisk box with the FWD service - I have the following in my sip.conf file: Have a look at http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions If you sip client is behind firewall you will not be able to connect to FWD. However you can get around by using IAXTEL. check out this page: www.iaxtel.com/setup.html David Kwok smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] [Fwd: reject connect from iaxtel.com]
I have just resolved the problem and included in this email, hoping it will help other people of get incoming Iaxtel working. Asterisk, at the moment, does not work with FWD. But Iaxtel and FWD can interconnect therefore having an Iaxtel account is the same as FWD account. I have read through a lot of google search and the materials are scare. I think it would be useful to document it. For Iaxtel to direct calls into your *, you will need to set up your iax.conf as follows: Under general section, the default will be fine. You need to register iax with iaxtel so that FWD user can contact you. [general] register: dkwok:[EMAIL PROTECTED] But you have to set up a client to allow Iaxtel to put calls through. [iaxtel] type=friend host=iaxtel.com context=from-iaxtel For type, type=friend * will allow both incoming and outgoing type=user * will allow incoming type=peers * will allow outgoing context will direct the incoming call to the context section in extensions.conf. In this case, [from-iaxtel]. You will need to setup exten under [from-iaxtel]. When Iaxtel sending through call it is in the format [EMAIL PROTECTED]/[EMAIL PROTECTED], therefore, you have to set up your context in extensions.conf as follows: [from-iaxtel] exten =s,1,Wait(1) exten =s,2,Answer exten =_.,3,Dial(sip/1001,20,tr) exten =_.,4,Hangup Have fun with *, thanks to Mark and his team. David Kwok ---BeginMessage--- Hi All I have problem trying to receive incoming calls from iaxtel.com. The error message is rejected connect from ip address - iaxtel.com. I have set up the iax.conf file as follow: port=5036 allow=gsm register=dkwok:[EMAIL PROTECTED] [dkwok] type=friend context=from_iaxtel My extensions.conf is as follows: [from_iaxtel] exten = 17001813482,1,Dial(sip/1001,20,tr) iax2 show registry 69.73.19.178:4569 dkwok 203.219.xxx.xxx:1200 60 Registered But when connection is attempted, the console says : File chan_iax2.c, Line 4301 (socket_read): Rejected connect attempt from 69.73.19.178. Any pointer will be appreciated. Also when iaxtel put call through local iax server does it send username password and is it matched up with the client section in iax.conf? David Kwok smime.p7s Description: S/MIME Cryptographic Signature ---End Message--- smime.p7s Description: S/MIME Cryptographic Signature
