similar to: WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'

Hi, Red Hat 9.0 Asterisk 1.2.7.1 I'm having a bit of an intermittent problem with my SIP account. Often (but not always) when I start * or RELOAD my dial plan from the CLI I get this message: >Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822 add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 >Apr 30 11:01:21 WARNING[12785]: acl.c:244

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

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

After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hello, I've been trying to setup a Fax2Email mecanism on my Asterisk box. I have been using the following: 1) An incoming IAX line on Unlimitel (Im not even sure if it's worth mentionning the provider, but I do just in case) 2) NVBackGroundDetect from Newman Telecom 3) The following extension to test: exten => fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

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, Does anyone know who did the prompts for French and Russian for Asterisk? I need some custom prompts. Regards, Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160511/ae5eea65/attachment.html>

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

I'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Last night I went to replace an Asterisk 1.4 + mISDN + b410p box with Asterisk 1.6.2 and DAHDI BRI - to no avail. I had two servers so copied network setting etc from the working one, moved the card across, ran dahdi_genconf etc and it didn't work. Here's the console output with notices disabled: lucas*CLI> pri intense debug span 1 Enabled debugging on span 1 [Dec 16

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

With 1.6.1 svn: [2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout: -- Registration for '17470121145 at proxy01.sipphone.com' timed out, trying again (Attempt #30) [2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'proxy01.sipphone.com' [2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register: Probably a DNS

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

Ottawa, Canada ? February 15, 2006 - PIKA Technologies Inc. today announced that they have integrated PIKA?s high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability

Hello, all, I have a Linksys 3102 from a VoIP provider. It use SRV record to register to the provider's SIP server. When I configure this line into my Asterisk, the register doesn't work if I use their domain name. So it like this: If I use register => user:pwd at proxy.provider.com then I got: [2010-05-17 11:47:19] WARNING[2366] chan_sip.c: No such host: proxy.provider.com

As I got to compile 1.0.6 and got it to run but having the same problem as before I thought of creating a new mail thread about this instead of continuing with one where topic is about something else. (Sorry) So, I can't do register anymore. It worked just a couple of days before and I haven't done anything special as far as I remember. *CLI outputs: Feb 28 22:16:55 NOTICE[53475]:

Does anyone have any recommendations for a SIP/IAX provider I can use for inbound callls? The plan is to have a 1800 number people can call and reach my Asterisk. The only provider I'm familiar with is Vonage, but they don't really like the idea of Asterisk according to their terms of service and they would probably notice if there are for example 3 calls coming in at the same time.

Hi, I'm looking for a new Internet Telephony Service Provider for my company in Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in Otawa, Toronto, NY & San Jose. Anyone out ther who can help me with a recommendation? Vonage seemed clueless when I called them. Broadvoice is good but no Canadian DIDs... Thanks, Hugh -------------- next part -------------- An