Thanks for the tip. I'll give that a try. I changed my DTMF settings, both on Asterisk and the Grandstream, to inband and it seems to be working now. I'll check out the delay the next time someone calls.
On Saturday 18 August 2007 8:23 pm, Philip Mullis wrote: > I see one possible issue, swap the order of the iax account codecs to ulaw > first then gsm or simply axe the gsm codec. This may correct your issue > right away. > > Regards, > Philip Mullis > > > > -----Original Message----- > From: Dave M. Sullivan on behalf of Dave Sullivan > Sent: Sat 8/18/2007 5:36 PM > To: [email protected] > Subject: Re: [on-asterisk] DTMF and delay issues > > Good idea... sorry ;) > > I'm running Asterisk 1.2.13 without a Zaptel timer (running on a virtual > dedicated server, so I can't load kernel modules unfortunately) and my home > phone goes through a Bell Sympatico HSE connection to Asterisk, which is > hosted on a Linode (linode.com). > > My VoIP provider is Voice Network (voicenetwork.ca) > > My ATA connects to Asterisk through SIP, and Asterisk connects to my VoIP > provider through IAX. > > Appropriate config entries: > > sip.conf > [home] > type=friend > context=user-dial > username=home > secret=******** > callerid=DAVE & ANGIE <6474763738> > nat=yes > qualify=yes > disallow=all > allow=ulaw > host=dynamic > canreinvite=no > > iax.conf > [voicenetwork_peer] > type=friend > host=did.voicenetwork.ca > dtmfmode=rfc2833 > insecure=very > disallow=all > allow=gsm > allow=ulaw > context=voicenetwork-in > > > If there's any more information I can provide, let me know. > Thanks! > > On Saturday 18 August 2007 5:09 pm, Remzi Semsettin Turer wrote: > > The first rule of debugging problems, provide as much info as you can. > > There is so many unknowns, such as: > > > > Asterisk version > > Connection speed/provider > > Voip provider > > Connection method (sip, iax) > > Your asterisk config (such as sip.conf) > > > > Then it would be so much easier to find root cause, instead of guess > > work. > > > > Cheers, > > > > Remzi > > > > ------------------------------------ > > Sent from my mail2web.com BlackBerry > > > > ----- Original Message ----- > > From: Dave Sullivan <[EMAIL PROTECTED]> > > To: [email protected] <[email protected]> > > Sent: Sat Aug 18 16:40:45 2007 > > Subject: [on-asterisk] DTMF and delay issues > > > > Hi All, > > > > I'm Dave, and I'm new to this mailing list. I just finished moving my > > home and business phones to VoIP using Asterisk, but there's two little > > issues that are somewhat annoying. > > > > The first is a DTMF problem. Whenever my girlfriend calls in to her work > > voicemail, she has problems entering her mailbox number and password. She > > enters it, but the remote system says her entry was invalid. I'm using a > > standard PSTN cordless phone with an ATA (Grandstream HT-101). Any > > suggestions? > > > > The other one, which I don't think anyone else has noticed yet, is that > > when someone calls us, it takes nearly a second before the other party > > can hear us (forcing us to say "hello?" twice all the time). I'm not sure > > where to begin diagnosing this one... can anyone offer any advice? > > > > Thanks in advance! > > > > -- > > Dave Sullivan > > [EMAIL PROTECTED] > > 647-235-0328 > > > > --------------------------------------------------------------------- > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > For additional commands, e-mail: [EMAIL PROTECTED] -- Dave Sullivan [EMAIL PROTECTED] 647-235-0328 --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
