Hi, After a long way of problems (shipping, customs, etc) finally I got Welltech working. Here below my comments. - The documentation is poor and have errors - The web configuration is not complete. However is useful for the basic configuration parameters. The command line is necessary for modify all parameters. - The software upgrade is easy. Initially the gw came with H323, we upgrade to SIP. - We have tested only one port, it works well, audio quality is good (alaw). - Outgoing and incoming calls are working ok. - The Caller ID (from PSTN side) does not work - Answer supervision (reversal polarity detection) seems to work fine. This feature is very important to us, is the first time that we found this feature in a analog CO trunk. In a test application where we play a voice message to the called user, the message start to play just after answer. Tested with wire phone and cell phones. - Disconnect tone seems reliable (although the default configuration was not adjusted). We have done dozen of test in order to get the gw working. During the tests two issues came up, they need further analysis and tests: - Two times a UDP packages loop between the gw and * saturated the bandwidth after a hung up. Rebooting the gw does not stop the loop. Even with the gw turn off, * was sending the packages.Only rebooting * turn the system normal. - The gw port stay locked after a hung up. Apparently due to a no detection of the disconnect tone (in this case the tests were carried out with a PABX without disconnect tone). But the * user (SIP) was hung up and it seems that there are not a release timer. We will continue the tests and test the Welltech technical support as well (no required until now). Jorge
I have a Welltech 3502 (2 FXS ports) and callerid will not work in SIP mode. I contacted Welltech support and they informed me that callerid is only working with the H.323 firmware. Once I flashed it with the H.323 firmware and figured out how to get it to work with asterisk, callerid did indeed start working. Joseph Tanner <joseph@thetechguide.com>> Message: 15 > Date: Fri, 02 Apr 2004 11:13:35 -0500 > From: Jorge Mendoza <mendoza@tcc.com.pe> > Organization: TCC S.A. > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Welltech FXO: initial tests > Reply-To: asterisk-users@lists.digium.com > > Hi, > > After a long way of problems (shipping, customs, etc) finally I got > Welltech working. Here below my comments. > > - The documentation is poor and have errors > - The web configuration is not complete. However is useful for the basic > configuration parameters. The command line is necessary for modify all > parameters. > - The software upgrade is easy. Initially the gw came with H323, we > upgrade to SIP. > - We have tested only one port, it works well, audio quality is good > (alaw). > - Outgoing and incoming calls are working ok. > - The Caller ID (from PSTN side) does not work > - Answer supervision (reversal polarity detection) seems to work fine. > This feature is very important to us, is the first time that we found > this feature in a analog CO trunk. In a test application where we play a > voice message to the called user, the message start to play just after > answer. Tested with wire phone and cell phones. > - Disconnect tone seems reliable (although the default configuration was > not adjusted). > > We have done dozen of test in order to get the gw working. During the > tests two issues came up, they need further analysis and tests: > - Two times a UDP packages loop between the gw and * saturated the > bandwidth after a hung up. Rebooting the gw does not stop the loop. Even > with the gw turn off, * was sending the packages.Only rebooting * turn > the system normal. > - The gw port stay locked after a hung up. Apparently due to a no > detection of the disconnect tone (in this case the tests were carried > out with a PABX without disconnect tone). But the * user (SIP) was hung > up and it seems that there are not a release timer. > > We will continue the tests and test the Welltech technical support as > well (no required until now). > > Jorge > > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest >
Hi, I'm using the Welltech pstn GW 3804 (four analogue ports) and in some way I agree with Jorge's points. I am also using two Welltech SIP Phone LAN 201 I set them in proxy mode. I am still left with some problems. I can talk between the two SIP phones only with reinvite (I cannot talk when * stays in the middle) I can call the outside pstn line through the GW, but I cannot hear the ringing tone (from the caller) and cannot speak. When I call from pstn, the gateway answer after the specified number of rings but it does not forward the call to the lan phone extension. I set the GW in peer to peer mode. I will attach the * config files and the welltech phone and gw configuration if needed. Any help is really appreciated. Claudio> Hi, > > After a long way of problems (shipping, customs, etc) finally I got > Welltech working. Here below my comments. > > - The documentation is poor and have errors > - The web configuration is not complete. However is useful for the basic > configuration parameters. The command line is necessary for modify all > parameters. > - The software upgrade is easy. Initially the gw came with H323, we > upgrade to SIP. > - We have tested only one port, it works well, audio quality is good (alaw). > - Outgoing and incoming calls are working ok. > - The Caller ID (from PSTN side) does not work > - Answer supervision (reversal polarity detection) seems to work fine. > This feature is very important to us, is the first time that we found > this feature in a analog CO trunk. In a test application where we play a > voice message to the called user, the message start to play just after > answer. Tested with wire phone and cell phones. > - Disconnect tone seems reliable (although the default configuration was > not adjusted). > > We have done dozen of test in order to get the gw working. During the > tests two issues came up, they need further analysis and tests: > - Two times a UDP packages loop between the gw and * saturated the > bandwidth after a hung up. Rebooting the gw does not stop the loop. Even > with the gw turn off, * was sending the packages.Only rebooting * turn > the system normal. > - The gw port stay locked after a hung up. Apparently due to a no > detection of the disconnect tone (in this case the tests were carried > out with a PABX without disconnect tone). But the * user (SIP) was hung > up and it seems that there are not a release timer. > > We will continue the tests and test the Welltech technical support as > well (no required until now). > > Jorge-- Claudio Loletti (Lollo) mailto:claudio.loletti@email.it jid: claudiolollo@jabber.org yahoo: lollonet2001@yahoo.it ICQ: 12096475 msn: claudiolol@hotmail.com GnuPG public key available on keyservers Key fingerprint = 40AB B2CB 5022 507B 5167 587C B1BA 90AC 6ECD 94D9 ---
Claudio.loletti
2004-Jun-16 10:40 UTC
[Asterisk-Users] [Asterisk-Users]Re: Welltech FXO: initial tests
Hi! this is the situation so far. the welltech 3804 is in peer mode, 2nddial is set to 2, the bureau table is pointing to extension 9 in extension.conf. we still have three problems! 1. if I call from outside the 3804 call ext 9 and I can hear the asterisk voice telling me to dial the extension required but everything I dial is not received from asterisk. 2. if I call from an internal sip phone to the pstn I get connected to the other side but I cannot hear the voice from pstn, while pstn can. 3. the voice between the internal sip phones is working only if reinvite is used (phone welltech lan 201 SIP). Jorge, can you please post your config of the 3804 so that I can compare it with mine? I tried to trace packets on the net in both configuration (with and without reinvite). in both of the tracing I found some ICMP reply from asterisk back to both the phones with type 3 and code 10 (destination unreachable). Isn\'t that strange? Thank you Claudio -- Email.it, the professional e-mail, gratis per te: http://www.email.it/f Sponsor: Da oltre 30 anni nel campo della didattica e della formazione Cepu ti offre una garanzia nella preparazione universitaria Clicca qui: http://adv.email.it/cgi-bin/foclick.cgi?mid=2613&d=20040616 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040616/5c1ebd35/attachment.htm
Hi Jorge! Our application rom version is 4fxosip.102 boot version is boot.104 I think we need to upgrade the app rom to version 103. I get into welltech ftp server and found a file called 4fxosipN2004_05_17.BIN. Do you know if that is the last version for the 3804? I solved some of the problems I had. 1. I can call between the 2 phones with and without reinvite. 2. I can call from SIP to pstn If I call from pstn, the 3804 answer and it dials extension 9 as specified in the bureau table, but it annot dial any internal extension. I hope to solve this last prob with the firmware upgrade Many thanks for you help Best Regards Claudio Loletti -- Email.it, the professional e-mail, gratis per te: http://www.email.it/f Sponsor: Lerboristeria.biz: per la tua bellezza e salute il miglior assortimento di prodotti erboristici ed oggettistica online Clicca qui: http://adv.email.it/cgi-bin/foclick.cgi?mid=2152&d=20040617 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040617/743253aa/attachment.htm
Claudio.loletti
2004-Jun-24 10:25 UTC
[Asterisk-Users] [Asterisk-Users]Re: Welltech FXO: initial tests
Hi! After flashing the rom to the new version and reconfiguring the unit, many problems were solved... Unfortunatly today something strange seems to happen. The 3804 stopped sending registration request to the asterisk proxy. Here it is the most significant part of its configuration: proxy mode proxy ip = asterisk ip sip port = 5060 (asterisk sip port is the same) RTP port 16384 expire = 900 security line1 account = pstn01 line1 pwd = test line2 account = pstn02 line2 pwd = test and so up to four system config keypad type = inband inter digi time = 3 ring time = 200 Ring before answer = 1 end of dial = disable voice setting 1st codec = A-law 2nd codec = u-law only those 2 are enbled on asterisk silence suppression = off echo cancelling = on phone book and prefix config are empty (default) routing table not changed (default) here is the result of the \"sip show peers\" cmd on the * console ns04*CLI> sip show peers Name/username Host Mask Port Status pstn04 (Unspecified) (D) 255.255.255.255 0 UNKNOWN pstn03 (Unspecified) (D) 255.255.255.255 0 UNKNOWN pstn02 (Unspecified) (D) 255.255.255.255 0 UNKNOWN pstn01 (Unspecified) (D) 255.255.255.255 0 UNKNOWN 6004 (Unspecified) (D) 255.255.255.255 0 UNKNOWN lan02/lan02 192.168.103.141 (D) 255.255.255.255 5060 OK (66 ms) lan01/lan01 192.168.103.140 (D) 255.255.255.255 5060 OK (64 ms) ns04*CLI> exit here is part of the sip.conf file [pstn01] context=local-calls type=friend host=dynamic md5secret=31389397dd34de1660d10f8da6d3bde7 ; secret=test dtmfmode=inband qualify=yes callerid=\"pstn01\" <065600001> defaultip=192.168.103.138 port=5060 canreinvite = yes here is the resul of the \"sip show users\" cmd n the * console ns04*CLI> sip show users Username Secret Authen Def.Context A/C pstn04 md5 local-calls No pstn03 md5 local-calls No pstn02 md5 local-calls No pstn01 md5 local-calls No 6004 local-calls No lan02 md5 local-calls No lan01 md5 local-calls No The strangest thing is that the 3804 at the very beginning, started to fail registration and after a couple of reboot stooped to send any REGISTRATION msg. Hope you can help! Thanks in advance Claudio -- Email.it, the professional e-mail, gratis per te: http://www.email.it/f Sponsor: Grandi offerte e vini pregiati, prova subito solo su Giordano vini! Clicca qui: http://adv.email.it/cgi-bin/foclick.cgi?mid=2623&d=20040624 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040624/ca47dbb7/attachment.htm
Caio Augusto Martimiano da Costa
2005-Jan-19 10:42 UTC
[Asterisk-Users] Welltech FXO: initial tests
Dear Claudio, I'm testing the welltech gateways (3804 firmware 4fxosip.102) and I am trying to make the Asterisk answer the calls from 3508 directly (with 2nddial off) it means throw hotline service. Do you know how to make the Asterisk answer a call from :pstn-to-3508 and 3508-hotline-Asterisk ? Please let me know if the question is not clear enough ! My configurations are: extension.conf: [general] static=yes writeprotect=no [default] include => oi exten => s,2,Answer exten => s,3,Wait(1) exten => s,4,Background(vm-toenternumber) ; Qual a extenss?o desejada? exten => 1,1,Goto(oi) exten => 2,1,Goto(oi) exten => 3,1,Goto(oi) exten => 6,1,Goto(oi) exten => 11,1,playback(beep) exten => 0,1,Goto(default,s,2) exten => t,1,Goto(timeout,s,1) exten => i,1,Goto,s|2 [oi] exten => s,1,Wait(1) exten => s,2,Answer exten => s,3,Wait(1) exten => s,4,playback(pbx-transfer) exten => s,5,Goto(default,s,2) [bogon-calls] exten => _.,1,Congestion sip.conf [general] port=5060 bindaddr=0.0.0.0 context=from-sip ;context=bogon-calls ;context=default maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=alaw [300] port=5060 type=friend context=default username=9 ; Username to use in INVITE until peer registers secret=fisa9 host=10.150.3.100 disallow=all allow=ulaw allow=alaw ;allow=g729 3804: usr/config$ sip -print Run Mode : PROXY MODE Proxy server address : 10.150.3.4 Domain : null Prefix string : 1234 Line1 : 100 Line2 : 101 Line3 : 102 Line4 : 103 SIP port : 5060 RTP port : 16384 Expire : 3600 usr/config$ sysconf -print System information Inter-Digit time out : 1 End of Dial : No end of dial Port status: port1: Enabled port2: Enabled port3: Enabled port4: Enabled DTMF selection : In-band RFC2833 Payload Type : 96 FAX Payload Type : 101 2nddial: 3 Billing: OFF Dial Rule ip side: filter: [] drop: [] insert: []. pstn side: filter: [] drop: [] insert: []. PIN prompt: 0 set1: 1111 set2: 2222 set3: 3333 set4: 4444 Ring Detect Method: 1 Ring before Answer: 0 usr/config$ bureau -print Bureau line setting relate information PSTN number : 4198 2000 2001 2002 Hold tone generation : On Hot line / Line to Line table ====================================================Port Destination Address Remote TEL/CHANNEL ----------------------------------------------------- 1 10.150.3.4 300 2 300 300 3 300 300 4 300 300 ========================================================= -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050119/ee9a949c/attachment.htm
Caio, Do you have the firmware files ?, I have a 3804 h323 and I'd like to upgrade it to SIP. The files are: - 2m4sipfxo.103 - 4fxosip.103 Kind regards, Miguel ------------------- From: "Caio Augusto Martimiano da Costa" <caio@furukawa.com.br> Subject: [Asterisk-Users] Welltech FXO: initial tests To: <asterisk-users@lists.digium.com> Message-ID: <s1ee8021.099@ctb-fisa5.furukawa.com.br> Content-Type: text/plain; charset="iso-8859-1" Dear Claudio, I'm testing the welltech gateways (3804 firmware 4fxosip.102) and I am trying to make the Asterisk answer the calls from 3508 directly (with 2nddial off) it means throw hotline service. Do you know how to make the Asterisk answer a call from :pstn-to-3508 and 3508-hotline-Asterisk ? Please let me know if the question is not clear enough ! My configurations are: extension.conf: [general] static=yes writeprotect=no [default] include => oi exten => s,2,Answer exten => s,3,Wait(1) exten => s,4,Background(vm-toenternumber) ; Qual a extenssco desejada? exten => 1,1,Goto(oi) exten => 2,1,Goto(oi) exten => 3,1,Goto(oi) exten => 6,1,Goto(oi) exten => 11,1,playback(beep) exten => 0,1,Goto(default,s,2) exten => t,1,Goto(timeout,s,1) exten => i,1,Goto,s|2 [oi] exten => s,1,Wait(1) exten => s,2,Answer exten => s,3,Wait(1) exten => s,4,playback(pbx-transfer) exten => s,5,Goto(default,s,2) [bogon-calls] exten => _.,1,Congestion sip.conf [general] port=5060 bindaddr=0.0.0.0 context=from-sip ;context=bogon-calls ;context=default maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=alaw [300] port=5060 type=friend context=default username=9 ; Username to use in INVITE until peer registers secret=fisa9 host=10.150.3.100 disallow=all allow=ulaw allow=alaw ;allow=g729 3804: usr/config$ sip -print Run Mode : PROXY MODE Proxy server address : 10.150.3.4 Domain : null Prefix string : 1234 Line1 : 100 Line2 : 101 Line3 : 102 Line4 : 103 SIP port : 5060 RTP port : 16384 Expire : 3600 usr/config$ sysconf -print System information Inter-Digit time out : 1 End of Dial : No end of dial Port status: port1: Enabled port2: Enabled port3: Enabled port4: Enabled DTMF selection : In-band RFC2833 Payload Type : 96 FAX Payload Type : 101 2nddial: 3 Billing: OFF Dial Rule ip side: filter: [] drop: [] insert: []. pstn side: filter: [] drop: [] insert: []. PIN prompt: 0 set1: 1111 set2: 2222 set3: 3333 set4: 4444 Ring Detect Method: 1 Ring before Answer: 0 usr/config$ bureau -print Bureau line setting relate information PSTN number : 4198 2000 2001 2002 Hold tone generation : On Hot line / Line to Line table ====================================================Port Destination Address Remote TEL/CHANNEL ----------------------------------------------------- 1 10.150.3.4 300 2 300 300 3 300 300 4 300 300 ==========================================================