Mark Raming
2004-Nov-07 04:06 UTC
[Asterisk-Users] Problem with call originating from Cisco 7940 SIP phone to a SIP peer
Hi, I'm having the following setup: Asterisk server (duh) running 1.0.0 stable release Cisco 7940 SIP phone. SIP peer to bridge to the PSTN world (www.pilmo.com) With the sip.config shown below, things work like a charm (changed phone numbers and passwords to protect the innocent). The Cisco uses the ULAW codec as can be expected, and the GSM codec is used with pilmo, so asterisk does the transcoding. However, when I remove the disallow=xxx and allow=gsm lines under the [pilmo] section, things become weird: I can still make outgoing calles. They end up using the a-law codec for both the Cisco and the pilmo channel. On the cisco, I can hear the calling/called party, but the calling/called party cannot hear me. I have also configured DIAX as a user. Making the same call from Diax (still using alaw with pilmo) works fine. Any clues? Am I doing something wrong? Is this a known problem for which I should get a more recent version of the code? Rgds, mark [general] context=external port=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw allow=alaw useragent=AddPac SIP Gateway nat=no register => 31165570909:hr0921dsf@nelson.ritstele.com:5060/31165570909 localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Incomming calls from Pilmo [212.26.192.155] type=user insecure=yes context=incomming ; Outgoing calls to Pilmo [pilmo] type=peer insecure=yes fromdomain=nelson.ritstele.com host=nelson.ritstele.com fromuser=31165570909 allow=gsm disallow=ulaw; disallow=alaw; dtmfmode=rfc2833 [2201] type=friend username=2201 secret=1234abcd! host=dynamic nat=never mailbox=2201 context=internal canreinvite=yes dtmfmode=rfc2833
Patrick
2004-Nov-07 06:29 UTC
[Asterisk-Users] Problem with call originating from Cisco 7940 SIP phone to a SIP peer
On Sun, 2004-11-07 at 12:06, Mark Raming wrote: [snip]> > ; Incomming calls from Pilmo > [212.26.192.155] > type=user > insecure=yes > context=incommingIt is usually spelled as "incoming". Typo or intentional?> ; Outgoing calls to Pilmo > [pilmo] > type=peer > insecure=yes > fromdomain=nelson.ritstele.com > host=nelson.ritstele.com > fromuser=31165570909 > allow=gsm > disallow=ulaw; > disallow=alaw; > dtmfmode=rfc2833The order is: 1) disallow=all 2) allow=... 3) allow=... etc. Remove: allow=gsm disallow=ulaw; <-- please note the ";" should not be there afaik disallow=alaw; <-- please note the ";" should not be there afaik Try with: disallow=all allow=gsm Regards, Patrick
Mark Raming
2004-Nov-09 02:16 UTC
[Asterisk-Users] Problem with call originating from Cisco 7940 SIP phone to a SIP peer
Hi, I have asked this same question earlier, but so far, I have not seen a solution yet... I'm having the following setup: Asterisk server running 1.0.0 stable release Cisco 7940 SIP phone. SIP peer to bridge to the PSTN world (www.pilmo.com) With the sip.config shown below, things work like a charm (changed phone numbers and passwords to protect the innocent). The Cisco uses the ULAW codec as can be expected, and the GSM codec is used with pilmo, so asterisk does the transcoding. However, when I remove the disallow=xxx and allow=gsm lines under the [pilmo] section, things become weird: I can still make outgoing calles. They end up using the a-law codec for both the Cisco and the pilmo channel. On the cisco, I can hear the calling/called party, but the calling/called party cannot hear me. I have also configured DIAX as a user. Making the same call from Diax (still using alaw with pilmo) works fine. Any clues? Am I doing something wrong? Is this a known problem for which I should get a more recent version of the code? Rgds, mark [general] context=external port=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw allow=alaw useragent=AddPac SIP Gateway nat=no register => 31165123456:password@nelson.ritstele.com:5060/31165123456 localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 localnet=169.254.0.0/255.255.0.0 ; Incomming calls from Pilmo [212.26.192.155] type=user insecure=yes context=incomming ; Outgoing calls to Pilmo [pilmo] type=peer insecure=yes fromdomain=nelson.ritstele.com host=nelson.ritstele.com fromuser=31165123456 disallow=ulaw disallow=alaw allow=gsm dtmfmode=rfc2833 [2201] type=friend username=2201 secret=1234abcd! host=dynamic nat=never mailbox=2201 context=internal canreinvite=yes dtmfmode=rfc2833
Patrick
2004-Nov-09 18:01 UTC
[Asterisk-Users] Problem with call originating from Cisco 7940 SIP phone to a SIP peer
On Tue, 2004-11-09 at 10:16 +0100, Mark Raming wrote:> [general] > context=external > port=5060 > bindaddr=0.0.0.0 > srvlookup=no > disallow=all > allow=ulaw > allow=alaw > useragent=AddPac SIP Gateway^^^^^^^^^^^^^^^^^ Is there a reason to change the useragent string? Maybe try commenting it out.> nat=no > register => 31165123456:password@nelson.ritstele.com:5060/31165123456 > localnet=192.168.0.0/255.255.0.0 > localnet=10.0.0.0/255.0.0.0 > localnet=172.16.0.0/12 > localnet=169.254.0.0/255.255.0.0 > > ; Incomming calls from Pilmo > [212.26.192.155] > type=user > insecure=yes > context=incommingIt is spelled "incoming". Typo?> ; Outgoing calls to Pilmo > [pilmo] > type=peer > insecure=yes > fromdomain=nelson.ritstele.com > host=nelson.ritstele.com > fromuser=31165123456 > disallow=ulaw > disallow=alaw > allow=gsmRemove the 2x disallow & 1x allow and try: disallow=all allow=gsm> dtmfmode=rfc2833 > > [2201] > type=friend > username=2201 > secret=1234abcd! > host=dynamic > nat=never > mailbox=2201 > context=internal > canreinvite=yes^^^^^^^^ So you want the Asterisk server to be taken out of the loop right? Regards, Patrick