Re: [sipx-users] Call Forwarding to External Number via PSTN not Working

2008-09-25 Thread Dale Worley
On Thu, 2008-09-25 at 10:40 -0700, IT Services wrote: > Hey there: > > I am configuring 3.10.2 with PSTN lines. I'm testing two gateways: > > Linksys 3102 > Grandstream 4108 > > Both gateways were virtually plug-and-play with respect to receiving > external calls and placing outbound calls. No p

Re: [sipx-users] Issue with remote registrations

2008-09-25 Thread Dale Worley
On Thu, 2008-09-25 at 22:53 +0300, James Mutuku wrote: > I am have an issue with remote registrations. With Xlite, I am getting > "error 503 - service unavailable", and there is no record of attempted > registrations in the sipregister.log files. You should check in the sipXproxy.log file to see

Re: [sipx-users] Basic SIP INVITE question...

2008-09-25 Thread Stephen D. Miller
Thanks for confirming, Scott. I knew you'd know ;-). Chalk up another apparent implementation issue for Grandstream gateways... sdm > -Original Message- > From: Scott Lawrence [mailto:[EMAIL PROTECTED] > Sent: Thursday, September 25, 2008 4:59 PM > To: Stephen D. Miller > Cc: Tony Gr

Re: [sipx-users] Basic SIP INVITE question...

2008-09-25 Thread Stephen D. Miller
Thanks Eric for the info. That's what I was looking for. It's a "low-cost" gateway (Grandstream GXW-4104) so I guess I shouldn't be that surprised. It works well for most configurations, but issues like this make me wish I'd simply moved a bit further up-market to a Patton or similar. sdm

Re: [sipx-users] Dialogic DMG Gateways

2008-09-25 Thread Tony Graziano
Without a call trace it's hard to tell, normally I think that maybe the DIALOGIC has a problem with REFER or hasn't been told explicitly how to handle it. I've seen that with gateways. What would be a good litmus test is to see if the AA works or if it drops the calls after the AA says "please

Re: [sipx-users] Basic SIP INVITE question...

2008-09-25 Thread Scott Lawrence
On Thu, 2008-09-25 at 15:59 -0400, Stephen D. Miller wrote: > > The "treat long distance prefix as optional" is indeed checked (and the > dialplan has been activated). > > That said, I think my core question is, according the applicable SIP > RFCs, is the "To:" field in the INVITE packet of S

[sipx-users] Dialogic DMG Gateways

2008-09-25 Thread Tony Wyland
Curious if anybody has had success with Dialogic series DMG gateways. We are experimenting with one and have one issue that seems likely to be caused by the gateway. First, what works --- calls to SipX extensions work fine, calls that go redirect to voicemail because there is no registered phone

Re: [sipx-users] Basic SIP INVITE question...

2008-09-25 Thread Stephen D. Miller
The "treat long distance prefix as optional" is indeed checked (and the dialplan has been activated). That said, I think my core question is, according the applicable SIP RFCs, is the "To:" field in the INVITE packet of SIP "allowed" to be different than the argument of the INVITE field? I

[sipx-users] Issue with remote registrations

2008-09-25 Thread James Mutuku
Hello List, I am have an issue with remote registrations. With Xlite, I am getting "error 503 - service unavailable", and there is no record of attempted registrations in the sipregister.log files. I used another softphone(sjphone) and there were records of attempted registrations in the sipr

Re: [sipx-users] Basic SIP INVITE question...

2008-09-25 Thread Tony Graziano
Good question. I don't think your issue is a gateway issue, rather go to your dialplan in questyion, click ADVANCED and see if the option: Treat long distance prefix as optional is checked, if not CHECK IT, apply, activate, try. Perhaps leave your proxy log set to debug and tail the log fi

[sipx-users] Basic SIP INVITE question...

2008-09-25 Thread Stephen D. Miller
In my long distance dial plan, I have sipX (v3.10) prepend the necessary 1 for dialing outbound long distance calls over the PSTN. Using a network sniff, I have verified that sipX does indeed do this and creates/sends the following INVITE to the gateway: INVITE sip:[EMAIL PROTECTED] SIP/

[sipx-users] Call Forwarding to External Number via PSTN not Working

2008-09-25 Thread IT Services
Hey there: I am configuring 3.10.2 with PSTN lines. I'm testing two gateways: Linksys 3102 Grandstream 4108 Both gateways were virtually plug-and-play with respect to receiving external calls and placing outbound calls. No problems there (except for low volume, but that's a different issue). Bu

Re: [sipx-users] problem with registrar

2008-09-25 Thread Scott Lawrence
On Thu, 2008-09-25 at 17:01 +0530, kavita gupta wrote: > hi , > all.my registrar server is failing again and again after some time.do > anyone have idea why it is doing so? > If anyone have idea please help me. Consider the following excellent advice when requesting help... http://www.chiark.g

Re: [sipx-users] sipXbridge - Request not issued from SIPX proxy server

2008-09-25 Thread Melcon Moraes
Sorry guys, This message should be posted only on sipx-dev list. My apologies. Cross-posting now. []'s MM 2008/9/25 Melcon Moraes <[EMAIL PROTECTED]>: > Hi guys, > > I'm a little bit confused. I was wondering if someone could help me > with: What exactly triggered the "Request not issued from

[sipx-users] receiving rfc 2833 DTMF event packets

2008-09-25 Thread Michael Socaciu
Trying again ; anybody using rfc 2833 can advise - thanks - mvs From: Michael Socaciu Sent: Monday, September 15, 2008 10:01 AM To: 'sipx-users@list.sipfoundry.org' Subject: receiving rfc 2833 DTMF event packets I am trying to set a DTMF digit

[sipx-users] sipXbridge - Request not issued from SIPX proxy server

2008-09-25 Thread Melcon Moraes
Hi guys, I'm a little bit confused. I was wondering if someone could help me with: What exactly triggered the "Request not issued from SIPX proxy server" condition? I know that this is new since r13492. I have my sipx on 192.168.22.254 and sipxbridge bound to 5081 for internal side and 5080 to ex

[sipx-users] problem with registrar

2008-09-25 Thread kavita gupta
hi , all.my registrar server is failing again and again after some time.do anyone have idea why it is doing so? If anyone have idea please help me. Kavita ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org

Re: [sipx-users] Sipx + spectralink phone

2008-09-25 Thread Pawel Kocik
Hi, Our problem is that when trying to make a call to spectralink 8002 or from spectralink 8002 (registered in sipx 3.10.2) we get error message "Assert: buffmgr.c Ln 1116" and then there is nothing else I can do apart from manual reboot of the phone. Polycom support said that they are

Re: [sipx-users] Sipx + spectralink phone

2008-09-25 Thread Vasilis Buklas
Do you have more info regarding this bug to contact polycom ? Thanks, Vasilis From: Pawel Pierscionek [mailto:[EMAIL PROTECTED] Sent: Thursday, September 25, 2008 11:43 AM To: Vasilis Buklas Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Sipx + spectralink phone O

Re: [sipx-users] Sipx + spectralink phone

2008-09-25 Thread Pawel Pierscionek
On 2008-09-25, at 07:48, Vasilis Buklas wrote: Hi to all, I have an installation of sipx 3.10 to my office and several spectralink 8020 wifi phones. I tried to configure them but while I managed to register the phones to the pbx when I initiate or accept a call from the phones Automati

Re: [sipx-users] SipX voicemail SIP URIs

2008-09-25 Thread Gabor Paller
Hi, I tried the "'sip:[EMAIL PROTECTED]" option and it is indeed the functionality we need. Thanks for the information. The second scenario would be the same as the scenario above except that the voicemail prompt would be skipped. I clarified with colleagues and that functionality is not critical

[sipx-users] Nortel 1535 Video

2008-09-25 Thread Gmb
Hi all, I try to use Nortel 1535 Video phone with sipx v.3.10.1 but the phone can't load configuration from ftp. I follow installation device procedure in sipx (Device -> Phones -> Add new Phones -> 1535 Video), generating profile, two files are created ( {MAC}.cfg and sysconf_2890d_sip.cfg ), b