similar to: no audio

I am trying to dial through my asterisk machine from phone A to phone B. My DID is registered properly with the SIP provider. When I dial from A to B it looks fine so far. A rings B and B can pick up and the call is bridged. However, I don't hear any audio so therefor it is not working. I am running Asterisk 1.8 on a cloud server. I have had the same configuration running on a physical

I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060

I am trying to use a simple tool in the Dial plan so that if the first number does not connect the logic will go to the second and/or third. Basically, I want the call to ring and connect to the first number Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

I have just started working with Asterisk and have run into a road block concerning IAX and an inbound DID from callwithus.com. I am getting nowhere and I don't really know how to isolate the problem. The asterisk version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can connect and make a call to other internal extensions using zoiper and iax. When I try and use the number,

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly hear music on hold and then it stops until the timeout for no answer

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

Calling from my home using Asterisk 1.6.2.1 to an office extension (Asterisk 1.6.1.13) the callerid is not honored: Home: -- Starting simple switch on 'DAHDI/1-1' -- Executing [170 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [170 at internal:2] NoOp("DAHDI/1-1", "Context: office-extensions") in new stack

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

Where is everyone located? I am in Washington DC. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/7c73847d/attachment.htm

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

Hello all, I know it was on the list before but i have some questions about the Kirk IP600v3, the requested configuration files were send private i guess Does anybody have the correct SIP settings for handsets connected to the Kirk. IP600v3 I am particulair intrested in settings regarding: -Voice Mailbox -Call waiting -DTMF settings for e.g. parking an extension with asterisk functionality

Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples.

Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic

I did an ngrep trace of what Asterisk realtime static does when it queries the database. Here's what I saw SELECT category, var_name, var_val, cat_metric FROM rt_pbx1_sip_vw WHERE filename='sip.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id; Firstly, why does it order in DESCENDING order by cat_metric? Shouldn't it be

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97

Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the "fixups" disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the

Does anyone know how to send a text message from Asterisk?