Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk CVS-HEAD-01/17/05-00:35:58 built by root@splurge on a i686 running Linux ==>SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all ; Allow all codecs allow=ulaw context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=1234 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050705/7a8d632b/attachment.htm
Do you have the open version or the Vonage one? Carlos Alperin Senior System Engineer Seneca Communications, LLC calperin@senecacom.net _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Walid Azab Sent: Tuesday, July 05, 2005 6:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk on Linksys WRT54G Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk CVS-HEAD-01/17/05-00:35:58 built by root@splurge on a i686 running Linux ==>SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all ; Allow all codecs allow=ulaw context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=1234 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050705/116d907e/attachment.htm
________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Walid Azab Sent: Tuesday, July 05, 2005 4:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk on Linksys WRT54G Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk CVS-HEAD-01/17/05-00:35:58 built by root@splurge on a i686 running Linux ==>SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all ; Allow all codecs allow=ulaw context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=1234 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) How are the routers connected to the IP network? Any nat before them on either end? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050705/fb2b788a/attachment.htm
Walid Azab wrote:> Hi all, > > Any one tried installing Asterisk on Linksys WRT54G? We have but facing > problems with SIP to SIP calls. The phones ring and calls are > established but we cannot hear any voice at all. I tried allow=all in > the general section but did not work. So I forced ulaw. Can any one > please check it out and let me know what is wrong? > > Here are the conf files: > > > *Asterisk Version:* Asterisk CVS-HEAD-01/17/05-00:35:58 built by > root@splurge <mailto:root@splurge> on a i686 running Linux > ** > *==>SIP.CONF* > ** > [general] > > port = 5060 ; Port to bind to (SIP is 5060) > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) > disallow=all ; Allow all codecs > allow=ulaw > context = bogon-calls ; Send SIP callers that we don't know about here > ** > ** > [2000] > > type=friend ; This device takes and makes calls > username=2000 ; Username on device > secret=1234 ; Password for device > host=dynamic ; This host is not on the same IP addr every time > context=from-sip ; Inbound calls from this host go here > mailbox=100 ; Activate the message waiting light if this > ; voicemailbox has messages in it > > [2001] ; Duplicate of 2000, except with different auth data > > type=friend > username=2001 > secret=1234 > host=dynamic > context=from-sip > mailbox=101 > > *==>Extensions.conf* > [general] > static=yes > writeprotect=yes > > [bogon-calls] > exten => _.,1,Congestion > > [from-sip] > exten => 2000,1,Dial(SIP/2000,20) > exten => 2000,2,Voicemail(u2000) > exten => 2000,102,Voicemail(b2000) > exten => 2000,103,Hangup > > exten => 2001,1,Dial(SIP/2001,20) > exten => 2001,2,Voicemail(u2001) > exten => 2001,102,Voicemail(b2001) > exten => 2001,103,Hangup > > exten => 2999,1,VoicemailMain(${CALLERIDNUM}) > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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
Sorry about that previous null reply. I was moving the window but hit the Send button by mistake :-( Walid Azab wrote:> Hi all, > > Any one tried installing Asterisk on Linksys WRT54G? We have but facing > problems with SIP to SIP calls. The phones ring and calls are > established but we cannot hear any voice at all. I tried allow=all in > the general section but did not work. So I forced ulaw. Can any one > please check it out and let me know what is wrong? > > Here are the conf files: > > > *Asterisk Version:* Asterisk CVS-HEAD-01/17/05-00:35:58 built by > root@splurge <mailto:root@splurge> on a i686 running Linux > **That's a pretty old version. The latest one on the site is: http://splurge.peoples-wireless.com/ipkg/asterisk-cvs_1.0.32_mipsel.ipk Which is CVS-head from just a couple of weeks ago. Are the phones which are trying to talk located on networks using NAT? That symptom is a classic one for SIP phones that are behind NAT. There are a variety of fixes, check out the Wiki at www.voip-info.org B.