I am trying to make a mirroring for my asterisk using nextone SBC,I have a problem ,which is when and end point send Invitation to SBC realm . This realm is send INV and REG messages to Asterisk. Asterisk sends INV message again to this realm. NexTone SBC try to send again to asterisk and this is caused loop. There solution was , Asterisk should send to a different realm of NexTone or different GW. How can I do that from asterisk.(define a signaling ip) Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. ********************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070529/9425e33b/attachment.htm
I am trying to make a mirroring for my asterisk using nextone SBC,I have a problem ,which is when and end point send Invitation to SBC realm . This realm is send INV and REG messages to Asterisk. Asterisk sends INV message again to this realm. NexTone SBC try to send again to asterisk and this is caused loop. There solution was , Asterisk should send to a different realm of NexTone or different GW. How can I do that from asterisk.(define a signaling ip) Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. ********************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070530/e82461f1/attachment.htm
BSumrall
2007-May-30 01:42 UTC
[asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!
after 18 hours, over 200 pages of reading, a complete reinstall of asterisk
I am down to this.
extensions.conf
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
[default]
exten => 8005181896,1,Dial,(IAX2/UXMC)
exten => s,1,Answer()
(I tried)
exten => _1XXXXXXXXXX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr)
(as well)
iax.conf
[general]
port=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
register => xxxx:xxxxxxxxxx@voip-co3.teliax.com
[teliax]
context=default
type=friend
host=voip-co3.teliax.com
auth=md5
user=xxxx
secret=xxxxxxxxx
disallow=all
allow=ulaw
allow=alaw
allow=gsm
sip.conf
[UXMC]
user=xxxxxxx
context=internal
type=friend
qualify=yes
nat=no
secret=xxxxxxxx
canreinvite=no
host=dynamic
nat=no
If I put back previous config, I can call into the 1800 number and here that
silly chick heckle me from my server!
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Jaswinder Singh
2007-May-30 02:53 UTC
[asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!
Can you post some output from asterisk cli output while you make call ? On 30/05/07, BSumrall <Brads@ftnco.com> wrote:> > > > > after 18 hours, over 200 pages of reading, a complete reinstall of asterisk > I am down to this. > > extensions.conf > > [globals] > CONSOLE=Console/dsp > IAXINFO=guest > TRUNK=Zap/g2 > TRUNKMSD=1 > > [default] > exten => 8005181896,1,Dial,(IAX2/UXMC) > exten => s,1,Answer() > > (I tried) > exten => _1XXXXXXXXXX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr) > (as well) > > iax.conf > > [general] > port=4569 > bandwidth=low > disallow=lpc10 > jitterbuffer=no > forcejitterbuffer=no > tos=lowdelay > autokill=yes > > register => xxxx:xxxxxxxxxx@voip-co3.teliax.com > > [teliax] > context=default > type=friend > host=voip-co3.teliax.com > auth=md5 > user=xxxx > secret=xxxxxxxxx > disallow=all > allow=ulaw > allow=alaw > allow=gsm > > sip.conf > > [UXMC] > user=xxxxxxx > context=internal > type=friend > qualify=yes > nat=no > secret=xxxxxxxx > canreinvite=no > host=dynamic > nat=no > > If I put back previous config, I can call into the 1800 number and here > that silly chick heckle me from my server! > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Eric "ManxPower" Wieling
2007-May-30 14:08 UTC
[asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!
You have too many codecs allowed. disallow=all and allow=ulaw in [general] and in each of the device sections of iax.conf. If that works, then you can start from there and try to get the codec you really want. BSumrall wrote:> after 18 hours, over 200 pages of reading, a complete reinstall of asterisk > I am down to this. > > extensions.conf > > [globals] > CONSOLE=Console/dsp > IAXINFO=guest > TRUNK=Zap/g2 > TRUNKMSD=1 > > [default] > exten => 8005181896,1,Dial,(IAX2/UXMC) > exten => s,1,Answer() > > (I tried) > exten => _1XXXXXXXXXX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr) > (as well) > > iax.conf > > [general] > port=4569 > bandwidth=low > disallow=lpc10 > jitterbuffer=no > forcejitterbuffer=no > tos=lowdelay > autokill=yes > > register => xxxx:xxxxxxxxxx@voip-co3.teliax.com > > [teliax] > context=default > type=friend > host=voip-co3.teliax.com > auth=md5 > user=xxxx > secret=xxxxxxxxx > disallow=all > allow=ulaw > allow=alaw > allow=gsm > > sip.conf > > [UXMC] > user=xxxxxxx > context=internal > type=friend > qualify=yes > nat=no > secret=xxxxxxxx > canreinvite=no > host=dynamic > nat=no > > If I put back previous config, I can call into the 1800 number and here that > silly chick heckle me from my server! > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >