sipsor+IPKall+GV no sound from caller (incoming/outgoing)

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slickdaddy96
Posts: 13
Joined: Fri Jan 20, 2012 8:25 pm

sipsor+IPKall+GV no sound from caller (incoming/outgoing)

Post by slickdaddy96 » Wed Feb 08, 2012 2:41 am

Ok I have had this problem that just started happening in the last day or two. I don't remember when it exactly started since I didn't use the phone yesterday or the day before I don't believe.

My issues is as follows whenever a call is placed (outgoing or incoming) it doesn't matter from my phone attached to my linksys SPA2102 I can never hear the other phone calling me or I'm calling. They can hear me, but I can't hear them. When dialing them I get the initial ring, but then it doesn't click to 2nd set of rings, it is just quiet, but if the people pick up they can hear me I just can't hear them. If I get an incoming call I don't get the "press 1 to accept call" from GV but if I press 1 it connects anyway, still can't hear them but they can hear me.

I have had someone call my IPKALL number and the same issue is going on so it isn't GV. I have reset the modem and the SPAP2102 several times and that is not the problem either. I have all the ports forwarded correctly because it was working fine previously. I am using a ruby simple dialplan. The only added addition to the dialplan that I have done recently is added callcentric as a SIP provider in sipsorcery and inserted that when 911 is called it uses callcentric instead of GV. I believe the phone was working fine after I did that insert in the dialplan also.

The preferred codec is G711u like it should be.

In the console everything completes like it was a successful call also with no errors.

Someone please help.

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by Aaron » Wed Feb 08, 2012 9:54 am

As a test add a rule to your dialplan that calls the music on hold number at iptel.org and see if you can hear the music.

Code: Select all

sys.Dial("music@iptel.org")

slickdaddy96
Posts: 13
Joined: Fri Jan 20, 2012 8:25 pm

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by slickdaddy96 » Wed Feb 08, 2012 1:30 pm

Aaron wrote:As a test add a rule to your dialplan that calls the music on hold number at iptel.org and see if you can hear the music.

Code: Select all

sys.Dial("music@iptel.org")
Forgive me I am not the most knowledgeable when it comes to dialplans. Good with tech stuff just bad in any kind of programming usually.

Where exactly in the dialplan should this be put, and what do I dial on my phone to get it to play the music, the numbers that spell M-U-S-I-C? I am using a landlined phone on an ATA not a softphone, so I am unsure on how I would dial that number.

Here is my dial plan for your reference:

# Copyright(c) 2010 Mike Telis

# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
# http://www.apache.org/licenses/LICENSE-2.0
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS, WITHOUT
# WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. See the
# License for the specific language governing permissions and limitations under
# the License.

# Click "View raw file" in the lower right for the best copy/paste view

AREA_CODE = 'xxx' # my area code
GV_USER = 'xxxg@gmail.com' # my GV e-mail address (user@gmail.com)
GV_PASS = 'xxx' # my GV password
CB_NUMBER = 'xxxx' # my 11-digit SIP number (only one)

SPEED_DIAL = { # my speed dial numbers
'1' => 'xxxx', # xxxx's Cell Phone
'2' => 'xxxx', # xxxxx's Cell Phone
'911' => '911@CallCentric', #911 through CallCentric
'411' => '17771234567@CallCentric', #CallCentric Info

}

begin
sys.Log "** Call from #{req.Header.From} to #{req.URI.User} **"

if sys.Out # if outbound call
num = req.URI.User.to_s # Get a string copy of the number to dial

num = SPEED_DIAL[num] || num # Substitute with speed dial entry, if any

case num
when /@/ then sys.Dial num # URI dialing
when /^[2-9]\d{6}$/ # Local call, 7-digit number
num = '1'+ AREA_CODE + num # prefix it with country and area code
when /^[01]?([2-9]\d{9})/ # US number with or without country code
num = '1' + $1 # add country code and truncate number to 10-digit
when /^(011|00|\+)(\d{9,})/ # international number
num = '+' + $2 # GoogleVoiceCall works with '+' prefix only


else sys.Respond 603, 'Wrong number, check & dial again'
end

sys.Log "Calling #{num} via Google Voice"
sys.GoogleVoiceCall GV_USER, GV_PASS, CB_NUMBER, num, '.*', CB_NUMBER =~ /^1747/ ? 7 : 1, 30

else # sys.Out
sys.Dial "#{sys.Username}@local"
end

rescue
sys.Log("** Error: " + $!) unless $!.to_s =~ /Thread was being aborted./
end

Also here what the console does when I try to make an outgoing call and the other phone answers obviously with all identifiable info xxx out:


DialPlan 13:40:45:222 sip1(5376): New call from tcp:xx.xx.xx.xxx:5074 successfully authenticated by digest.
DialPlan 13:40:45:238 sip1(5376): Using dialplan default for Out call to sip:xxxxxxxxxx@sipsorcery.com.
NewCall 13:40:45:253 sip1(5376): Executing script dial plan for call to xxxxxxxxxx.
DialPlan 13:40:45:300 sip1(5376): ** Call from "xxx" <sip:xxxxxxxxxx@sipsorcery.com>;tag=5d0b1f98d77c20efo0 to xxxxxxxxxx **
DialPlan 13:40:45:300 sip1(5376): Calling xxxxxxxxxxx via Google Voice
DialPlan 13:40:45:300 sip1(5376): SDP on GoogleVoiceCall call had RTP socket mangled from 192.168.1.8:16400 to xx.xx.xx.xxx:16400.
DialPlan 13:40:45:300 sip1(5376): UAS call progressing with Ringing.
DialPlan 13:40:45:300 sip1(5376): Logging into google.com for xxxxxxxxxxxx@gmail.com.
DialPlan 13:40:45:363 sip1(5376): Google Voice pre-login page loaded successfully.
DialPlan 13:40:45:394 sip1(5376): GALX key _o7t5WmJds0 successfully retrieved.
NATKeepAlive 13:40:45:535 sip1(5856): Requesting NAT keep-alive from proxy socket tcp:69.59.142.213:5060 to tcp:xx.xx.xx.xxx:5074.
DialPlan 13:40:46:269 sip1(5376): Google Voice home page loaded successfully.
DialPlan 13:40:46:300 sip1(5376): Call key nBN7Hfw24L8DA7ID/xxiDn8fHcs= successfully retrieved for xxxxxxxxxx@gmail.com, proceeding with callback.
DialPlan 13:40:46:316 sip1(1924): SIP Proxy setting application server for next call to user slickdaddy96 as udp:69.59.142.213:5070.
DialPlan 13:40:46:472 sip1(5376): Google Voice Call to xxxxxxxxxxx initiated, callback #xxxxxxxxxxx, phone type 1, timeout 30s.
DialPlan 13:40:48:191 sip1(1924): SIP Proxy directing incoming call for user xxxxxxxxxx to application server udp:69.59.142.213:5070.
DialPlan 13:40:48:191 sip1(5376): Google Voice Call callback received.
DialPlan 13:40:48:191 sip1(5376): Answering client call with a response status of 200.
DialPlan 13:40:48:285 sip1(5376): Google Voice Call was successfully answered in 2.98s.
DialPlan 13:40:48:285 sip1(5376): Dialplan cleanup for xxxxxxxxxx.
DialPlan 13:40:48:535 sip1(5376): Dial plan execution completed with normal clearing.
DialPlan 13:41:13:020 sip1(5376): Matching dialogue found for BYE to sip:69.59.142.213:5060;transport=tcp from udp:69.59.142.213:5060.

DoDo
Posts: 182
Joined: Wed May 20, 2009 7:16 pm

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by DoDo » Thu Feb 09, 2012 3:48 am

Read and Follow the process on my posts.
Here is The Link :
viewtopic.php?f=6&t=3857&p=21033#p21033


( When you are done with the Process just do this : Delete Your Sip Account then Add it again your Sip Account, Then Reboot your Ata ) ( on the Out Dial Plan Add Ipkall Number instead of Sipgate Number ).

slickdaddy96
Posts: 13
Joined: Fri Jan 20, 2012 8:25 pm

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by slickdaddy96 » Thu Feb 09, 2012 7:01 am

DoDo wrote:Read and Follow the process on my posts.
Here is The Link :
viewtopic.php?f=6&t=3857&p=21033#p21033


( When you are done with the Process just do this : Delete Your Sip Account then Add it again your Sip Account, Then Reboot your Ata ) ( on the Out Dial Plan Add Ipkall Number instead of Sipgate Number ).
That is all well and good but those "processes" don't have anything to do with my problem I don't have a sipgate.com account, and ipkall is a forwarding only service that forwards to sipsorcery.com, so there is no need to put them in the Sip Providers list: viewtopic.php?f=7&t=3888#p21171 See this post as a reference on that. Everything else in your process i already have setup, and again it was all working with no problem, nothing was changed then in the last few days I can't hear the callers if they call me or if I call them. The calls complete just fine, I just can't hear them, so it isn't an error that your processes would fix. Any other suggestions.

slickdaddy96
Posts: 13
Joined: Fri Jan 20, 2012 8:25 pm

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by slickdaddy96 » Thu Feb 09, 2012 7:22 am

By doing nothing but letting it sit, now I can hear the caller when the call is incoming and right before I tried incoming I tried outgoing and it worked once, but since then it is again when outgoing I can't hear the other side but they can hear me, but it looks as if the no sound incoming problem is fixed and it fixed on its on. Now if I can 'just figure out why outgoing is messed up still.

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by Aaron » Thu Feb 09, 2012 9:42 am

Are you able to get hold of a friend's modem/router to do a test with? It sounds like yours is behaving very erratically. One-way and no-audio issues are almost always caused by the NAT implementations on routers doing weird things.

Alternatively you could see if there is a firmware update for your own router or search around on the internet for anyone else who has had audio issues with the same router model.

slickdaddy96
Posts: 13
Joined: Fri Jan 20, 2012 8:25 pm

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by slickdaddy96 » Thu Feb 09, 2012 3:44 pm

I am running DD-WRT and must have been using a more unstable version for my router versus the reccommended completely stable version. So I flashed the router to the known most stable version, and everything works great now. So it was the router. Hopefully it will continue to work fine.

slickdaddy96
Posts: 13
Joined: Fri Jan 20, 2012 8:25 pm

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by slickdaddy96 » Fri Feb 10, 2012 8:00 pm

Ok the problem is back so it wasn't the firmware. I have three routers in the house all either WRT310n or E2000 Linksys/Cisco routers running DD-wrt. I have two bridges and the router that everything routes to. I have switched each of my routers as the main one and nothing fixes the problem. Here is a screenshot of port forwarding.
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Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Re: sipsor+IPKall+GV no sound from caller (incoming/outgoing

Post by Aaron » Fri Feb 10, 2012 11:37 pm

Are you able to place your PAP2 directly behind your first router? Trying to get RTP connected through two or more NATs is notoriously difficult.

If you can reduce your PAP2's network path to the simplest possible one it may help. My advice would be to connect your PAP2 directly to your internet facing router and if that still doesn't solve the problem then remove all your port forwarding rules; if your port forwarding rules are incorrect that will also cause big problems so initially its best to try and do without them.

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