similar to: Intermittent problem dialling out on a SIP channel

Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel: >-- Executing Dial("Zap/1-1", "SIP/6137451576@6477235412||t|") in new stack >Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such host: 6477235412 >Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create >channel of type

I've had a heck of a time getting a SIP channel to work in Asterisk 1.2.7.1 (Redhat 9.0). I've done it successfully a number of times on pre 1.2 versions of Asterisk so perhaps it's version related. Any insights are welcome! At first I wasn't able to dial out on the SIP channel _whenever_ I started Asterisk (i.e. not just when the box was booted). I always had to do a reload

Thanks. I know that line quality is a factor, and I know I could get a 50$ fax with a PSTN line (that is what I have now). But I have my reasons to want to setup a fax over IP, and I want to keep going. Where do I find info on this debug mode? Is there a detaild log in Asterisk that show exactly what happens when the fax is trying to come in? Also, could this console output help? - Executing

Hi, We are using Asterisk 1.6.2 and it is continually failing to resolve Verizon SRV and sending following message, WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' DNS settings on OS level is working fine. Can anyone have an idea about it? Regards, Faisal Hanif

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite

Hi all! So far I've always used plaintext passwords for SIP, but now I've decided to use MD5 encryption. For each client I edited its section as follows, then: auth=md5 md5secret=hashed_passwd ;secret=plaintext_passwd where hashed_passwd is the output of echo -n "user:realm:plaintext_passwd" | md5sum When the first SIP clients registers with Asterisk after a "sip

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello everybody, one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco <---pri---> asterisk <---pri---> legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

Hi all. I am seeing incoming calls from digital lines (mobiles e.g.) dialling my main number + 3-digit extension just fine ("Accepting voice call from '11234567' to '250' on channel 0/1, span 1"). The problem however is with calls from analog lines: "Accepting voice call from '13331846' to '25' on channel 0/1, span 1" * just sees 2 digits, not

Hello, I'm experimenting with Overlap Dialling in asterisk 1.6. I've enabled this in sip.conf and on the SNOM 300 phone. My problem is that asterisk dials out as soon as it matches an extension without waiting to see if the user is going to type in more digits. Is there a way to set a timeout per channel or globally? I'd like Asterisk to wait for a few seconds once its found a match

hi there was wondering if anybody knows this.. have successfully installed shady dial and the agent is now logging in successfully i've enabled postgres debugging and i found out that no request had been made by the shady dialer to query the database for the numbers... also i am able to login to the queue simply by entering the agent id... it doesnt ask for the password...it simply plays the

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and

hi asterisk users How can i make asterisk "ignore" invalid extensions, and go on playing the background soundfile? Normally, asteriks will take the user to the invalid extension if the caller presses anything other than 1 or 2 in the following context:: [example] exten => s,1,Answer() exten => s,2,Background(hello-world) exten => s,n,Goto(s,2) exten =>

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

Hi, I am getting this problem when trying to register with Voipfone.co.uk. It used to work, and I haven't changed anything that I know of. Jan 13 10:22:37 WARNING[21645]: acl.c:213 ast_get_ip_or_srv: Unable to lookup 'voipfone.co.uk.voipfone.co.uk' Why does the domain name appear twice? I don't know when it stopped working. In SIP.CONF [sip_proxy-out] type=peer

When trying to dial out 982420173 our main number I get the engaged signal before I finish entering the phone number Any ideas ???? Regards Mick

When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks!