On Wednesday 05 May 2004 10:42 pm, Kyle Hagan wrote:> Quick update. > > Still having problems.. But here is whats happening. > > When I call into the ZIP4x4 from anywhere (ZAP or SIP) it works fine. > > When I make ANY call from the ZIP4x4 to anywhere no audio.. Even if im > calling the switch to get into Voicemail. I see the * system playing the > audio on the console but I hear no audio and it doesnt accept DTMF tones > after the initial dial. > > If anyone has any ideas please let me know.Maybe a firewall problem? Anon
Having a problem, and have searched archives. I have no problems call from Softphones to ZAP or other soft phones. I just got a Zultys ZIP 4x4 in and can call to Soft phones but do get audio eitherway. If I call from Soft phone to the ZIP 4x4 hard phone it works fine. I get the errors below when calling from ZIP to Softphone. May 5 13:53:14 WARNING[1133718080]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on call 2363984802-110 for seqno 105 (Non-critical Request) May 5 13:53:17 WARNING[1133718080]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on call 2363984802-110 for seqno 105 (Critical Response) Also: When I dial 9 from the ZIP or dial *55 for voicemail it doesnt work. But if I dial the ext of a zap or sip it will ring fine. Please help Kyle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040505/c1ab1664/attachment.htm
Quick update. Still having problems.. But here is whats happening. When I call into the ZIP4x4 from anywhere (ZAP or SIP) it works fine. When I make ANY call from the ZIP4x4 to anywhere no audio.. Even if im calling the switch to get into Voicemail. I see the * system playing the audio on the console but I hear no audio and it doesnt accept DTMF tones after the initial dial. If anyone has any ideas please let me know. Kyle ----- Original Message ----- From: Kyle Hagan To: asterisk-users@lists.digium.com Sent: Wednesday, May 05, 2004 1:52 PM Subject: [Asterisk-Users] No Audio from Hard Phone to SIP Having a problem, and have searched archives. I have no problems call from Softphones to ZAP or other soft phones. I just got a Zultys ZIP 4x4 in and can call to Soft phones but do get audio eitherway. If I call from Soft phone to the ZIP 4x4 hard phone it works fine. I get the errors below when calling from ZIP to Softphone. May 5 13:53:14 WARNING[1133718080]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on call 2363984802-110 for seqno 105 (Non-critical Request) May 5 13:53:17 WARNING[1133718080]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on call 2363984802-110 for seqno 105 (Critical Response) Also: When I dial 9 from the ZIP or dial *55 for voicemail it doesnt work. But if I dial the ext of a zap or sip it will ring fine. Please help Kyle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040505/f3c340f6/attachment.htm
No firewall. Its all internal to our lan. Kinda feeling its a codec problem some how when the ZIP4x4 connects to asterisk. Kyle ----- Original Message ----- From: "Anon" <asterisk_user@tarottoni.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, May 05, 2004 11:48 AM Subject: Re: [Asterisk-Users] No Audio from Hard Phone to SIP> On Wednesday 05 May 2004 10:42 pm, Kyle Hagan wrote: > > Quick update. > > > > Still having problems.. But here is whats happening. > > > > When I call into the ZIP4x4 from anywhere (ZAP or SIP) it works fine. > > > > When I make ANY call from the ZIP4x4 to anywhere no audio.. Even if im > > calling the switch to get into Voicemail. I see the * system playing the > > audio on the console but I hear no audio and it doesnt accept DTMF tones > > after the initial dial. > > > > If anyone has any ideas please let me know. > Maybe a firewall problem? > > Anon > > _______________________________________________ > 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 >
You tried the standard disallow=all and allow=ulaw in sip.conf, right? On Thu, 2004-05-06 at 10:54, Kyle Hagan wrote:> No firewall. Its all internal to our lan. > > Kinda feeling its a codec problem some how when the ZIP4x4 connects to > asterisk.-- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."
have: [101] type=friend secret=123456 auth=md5 nat=yes host=dynamic reinvite=no canreinvite=no qualify=1000 dtmfmode=inband callerid="SIP Phone" <101> mailbox=101 disallow=all ;allow=gsm allow=alaw allow=ulaw context=home ----- Original Message ----- From: "Eric Wieling" <eric@fnords.org> To: <asterisk-users@lists.digium.com> Sent: Thursday, May 06, 2004 9:19 AM Subject: Re: [Asterisk-Users] No Audio from Hard Phone to SIP> You tried the standard disallow=all and allow=ulaw in sip.conf, right? > > On Thu, 2004-05-06 at 10:54, Kyle Hagan wrote: > > No firewall. Its all internal to our lan. > > > > Kinda feeling its a codec problem some how when the ZIP4x4 connects to > > asterisk. > > -- > Eric Wieling * BTEL Consulting * 504-899-1387 x2111 > "In a related story, the IRS has recently ruled that the cost of Windows > upgrades can NOT be deducted as a gambling loss." > > _______________________________________________ > 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 >
Allow ULAW or ALAW, not both, at least for trying to solve a problem. On Thu, 2004-05-06 at 11:36, Kyle Hagan wrote:> have: > > [101] > type=friend > secret=123456 > auth=md5 > nat=yes > host=dynamic > reinvite=no > canreinvite=no > qualify=1000 > dtmfmode=inband > callerid="SIP Phone" <101> > mailbox=101 > disallow=all > ;allow=gsm > allow=alaw > allow=ulaw > context=home > > > > ----- Original Message ----- > From: "Eric Wieling" <eric@fnords.org> > To: <asterisk-users@lists.digium.com> > Sent: Thursday, May 06, 2004 9:19 AM > Subject: Re: [Asterisk-Users] No Audio from Hard Phone to SIP > > > > You tried the standard disallow=all and allow=ulaw in sip.conf, right? > > > > On Thu, 2004-05-06 at 10:54, Kyle Hagan wrote: > > > No firewall. Its all internal to our lan. > > > > > > Kinda feeling its a codec problem some how when the ZIP4x4 connects to > > > asterisk. > > > > -- > > Eric Wieling * BTEL Consulting * 504-899-1387 x2111 > > "In a related story, the IRS has recently ruled that the cost of Windows > > upgrades can NOT be deducted as a gambling loss." > > > > _______________________________________________ > > 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 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-- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Eric Wieling wrote: | Allow ULAW or ALAW, not both, at least for trying to solve a problem. What is the difference between these codecs? Which is better? - -- Jason A. Pattie pattieja@xperienceinc.com Xperience, Inc. (http://www.xperienceinc.com) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAm7afuYsUrHkpYtARAqtZAKCAro/yoLCuRUjkuV8H2IKiJAbXBQCeMBSX Wz6h4cY7nTH2AoEAkqRPftY=XJTT -----END PGP SIGNATURE----- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support.
The manufacturer think its a bad phone. So Im getting another one today. They said thay have gotten the Zultys phones to work with asterisk with no problems. Will let everyone know. Zultys sait ULAW was the most common but did not state why most use it. Kyle ----- Original Message ----- From: "Jason A. Pattie" <pattieja@pcxperience.com> To: <asterisk-users@lists.digium.com> Sent: Friday, May 07, 2004 9:17 AM Subject: Re: [Asterisk-Users] No Audio from Hard Phone to SIP> -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Eric Wieling wrote: > | Allow ULAW or ALAW, not both, at least for trying to solve a problem. > > What is the difference between these codecs? Which is better? > > - -- > Jason A. Pattie > pattieja@xperienceinc.com > Xperience, Inc. (http://www.xperienceinc.com) > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.2.4 (GNU/Linux) > Comment: Using GnuPG with Debian - http://enigmail.mozdev.org > > iD8DBQFAm7afuYsUrHkpYtARAqtZAKCAro/yoLCuRUjkuV8H2IKiJAbXBQCeMBSX > Wz6h4cY7nTH2AoEAkqRPftY> =XJTT > -----END PGP SIGNATURE----- > > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > MailScanner thanks transtec Computers for their support. > > _______________________________________________ > 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 >