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Generate Your SIP Trunk Config: PTCL, Nayatel, Cybernet on Grandstream, Yeastar, 3CX or FreePBX

Enter your Pakistani ISP trunk parameters to generate a ready-to-paste PBX configuration with verified NAT traversal settings, G.711a codec precedence, and Pakistani dial plans. Eliminates one-way audio, 32-second drops, and SIP 488 rejections without credentials ever leaving your machine.

Supported Pakistani ISP & PBX Compatibility Matrix

Field-tested SIP trunk parameters across major Pakistani telecommunications carriers and IP PBX appliances.

Carrier / ISP Default Signaling Mandatory Codec Supported PBXs NAT Sensitivity
PTCL SIP Trunk UDP 5060 (GPON VLAN) G.711a (PCMA) Mandatory Grandstream UCM, Yeastar, 3CX, FreePBX High (Bridge mode required on ONT)
Nayatel Metro Fiber UDP 5060 / IP Peering G.711a, G.729 Grandstream, Yeastar P/S, Asterisk Medium (Static IP WAN)
Cybernet / StormFiber UDP 5060 (Peer/REGISTER) G.711a (PCMA 64k) Grandstream, Yeastar, 3CX v18/v20 Medium (Disable SIP ALG)
Transworld (TES) UDP 5060 (Metro Ethernet) G.711a, G.729 All RFC3261 PBX Systems Low (Dedicated VLAN)
Wateen Telecom UDP 5060 (PRI-to-SIP) G.711a (RFC2833) Grandstream UCM, Asterisk chan_pjsip Medium
Zong Business UDP 5060 (Fixed Wireless) G.711a, AMR Grandstream, FreePBX, 3CX Medium

Pakistani Carrier Technical SIP Profiles

Detailed registrar IP conventions, port assignments, codec hierarchies, and known carrier quirks.

PTCL SIP trunk settings

  • Registrar / SBC: Internal IMS SBC IP address (typically 10.240.x.x) routed via GPON optical VLAN 20/30.
  • Port & Transport: 5060 UDP. Keep-alive registration expiry must be set to 180 seconds or lower.
  • Codec Precedence: G.711a (PCMA) must be listed strictly in 1st position. G.711u or Opus will trigger SIP 488 (Not Acceptable Here).
  • Known Quirks: PTCL Huawei HG8245 & ZTE F660 GPON ONTs enforce carrier double-NAT and have SIP ALG enabled by default, corrupting SDP headers and causing 30-second one-way audio drops. Request bridge mode on Port 1/GE from PTCL NOC (1218).

Nayatel SIP trunk settings

  • Registrar: sip.nayatel.com or allocated corporate static gateway IP.
  • Port & Transport: 5060 UDP. Supports both SIP REGISTER and Peer Trunk (IP whitelisting).
  • Codecs: G.711a (PCMA) and G.729. DTMF delivery: RFC2833 payload 101.
  • Known Quirks: Trunks delivered via secondary optical media converter ports require a static route configured on your PBX when operating dual network interfaces.

Cybernet / StormFiber SIP trunk settings

  • Registrar: 175.107.x.x or allocated StormFiber enterprise SBC gateway.
  • Port & Transport: 5060 UDP. Expiration 180 seconds.
  • Codecs: G.711a (PCMA 64k) primary. G.729 secondary.
  • Known Quirks: StormFiber fiberhome/ZTE routers enable SIP ALG by default on port 5060. Disable under router admin Application > ALG > SIP ALG.

Transworld SIP trunk settings

  • Registrar: sip.tw1.com or enterprise direct BGP peering IP.
  • Port & Transport: 5060 UDP. Expiry 1800s.
  • Codecs: G.711a, G.729.
  • Known Quirks: Ensure PBX NAT traversal is disabled when the PBX has a public IP directly bound to its WAN Ethernet port.

Wateen SIP trunk settings

  • Registrar: sip.wateen.com or optical PRI gateway SBC.
  • Port & Transport: 5060 UDP. Expiry 1800s.
  • Codecs: G.711a with RFC2833 DTMF.
  • Known Quirks: Legacy PRI-to-SIP migrations may require caller ID manipulation in E.164 (+92) format.

Zong Business SIP trunk settings

  • Registrar: sip.zong.com.pk or corporate fixed LTE/5G voice gateway.
  • Port & Transport: 5060 UDP. Expiry 180s.
  • Codecs: G.711a, AMR, G.729.
  • Known Quirks: Disable SIP session timers or set to 1800s if calls terminate at exactly 15 minutes.

PBX Platform Configuration Standards

Field layouts, templates, and snippet samples for leading enterprise PBX systems.

Grandstream UCM SIP trunk configuration

On Grandstream UCM6200 and UCM6300 series appliances, navigate to Extension / Trunk > VoIP Trunks > Create SIP Trunk. Set Type to "Register SIP Trunk" (or "Peer SIP Trunk" if using static IP authentication). Configure Host Name with the carrier registrar IP, Transport to UDP, DTMF to RFC2833, and prioritize PCMA in Codecs. Under PBX Settings > SIP Settings > NAT, set NAT Mode to External IP and declare your RFC1918 subnets. 

# Grandstream UCM Importable CSV Header Sample
Trunk Name,Type,Provider,Host Name,Port,Transport,Username,Auth ID,Caller ID,Keepalive,DTMF,Codecs,NAT Mode
"PTCL_Trunk","register","ptcl","10.240.1.1","5060","UDP","02135891122","02135891122","02135891122","180","RFC2833","PCMA;G729;PCMU","external_ip"

Yeastar P-Series and S-Series SIP trunk configuration

In Yeastar P-Series and S-Series web managers, navigate to PBX > Trunks > Add Trunk. Configure Trunk Type, Hostname/IP, Port 5060, and Outbound Caller ID. Enable Qualify with a frequency of 180 seconds. Under System > PBX Settings > SIP > NAT, set NAT Type to "External IP Address" and declare local subnets (192.168.0.0/16, 10.0.0.0/8, 172.16.0.0/12). 

[Yeastar Trunk Deployment]
Trunk Type = Register Trunk
Hostname / IP = 10.240.1.1 | Port = 5060 | Transport = UDP
Qualify = Yes | Qualify Frequency = 180s | Codecs = PCMA, G729
NAT Type = External IP Address | Local Subnets = 192.168.0.0/16, 10.0.0.0/8

3CX Phone System SIP trunk configuration

In 3CX Management Console (v18) or Admin Console (v20), add a Generic SIP Trunk. Enter Main Trunk No (pilot DID), Registrar IP, and concurrent calls. In Options, rank G.711 A-law (PCMA) as #1 and disable Re-Invite. Important: 3CX utilizes RTP audio ports 9000–10999 UDP by default. 

<?xml version="1.0" encoding="utf-8"?>
<doc xmlns:tcx="http://www.3cx.com">
  <connection>
    <registrar>10.240.1.1</registrar><port>5060</port><transport>udp</transport>
    <simultaneouscalls>16</simultaneouscalls>
  </connection>
  <codecs><codec rank="1">PCMA</codec><codec rank="2">G729</codec></codecs>
</doc>

FreePBX & Asterisk chan_pjsip configuration

FreePBX 16/17 and Asterisk chan_pjsip require five linked sections: transport-udp, registration, auth, aor, endpoint, and an identify section matching incoming INVITEs by carrier registrar IP. Outbound dial rules should match 03XXXXXXXXX (mobile) and 02XXXXXXXXX (landlines). 

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
local_net=192.168.0.0/16
external_media_address=YOUR_WAN_IP
external_signaling_address=YOUR_WAN_IP

[ptcl_trunk]
type=endpoint
transport=transport-udp
context=from-pstn
disallow=all
allow=alaw
allow=g729
direct_media=no
force_rport=yes
rewrite_contact=yes
rtp_symmetric=yes

VoIP NAT Traversal & One-Way Audio Checklist

Prevent silent calls, one-sided voice packets, and 30-second disconnects across Pakistani routers.

  1. External Static WAN IP Declared: Configure your office border router's public static IPv4 address in the PBX SIP NAT settings so external SDP responses do not advertise RFC1918 private addresses.
  2. Local Subnets Declared: Declare 192.168.0.0/16, 10.0.0.0/8, and 172.16.0.0/12 as local networks so local deskphones are not subjected to external NAT translation.
  3. RTP Audio Port Forwarding: Forward UDP ports 10000–20000 (Grandstream/Asterisk/Yeastar) or 9000–10999 (3CX) from your border router directly to the PBX LAN IP.
  4. Disable Router SIP ALG:
    • PTCL GPON ONT (Huawei HG8245 / ZTE F660): Web Admin (192.168.1.1) > Security/Application > ALG Configuration > Uncheck SIP ALG. Call PTCL NOC (1218) to request Bridge WAN Mode on Port 1 to eliminate double-NAT.
    • TP-Link (Archer / Omada): Advanced > NAT Forwarding > ALG > SIP ALG: Disabled.
    • MikroTik RouterOS: Run in terminal: /ip firewall service-port disable sip
    • Fortinet FortiGate: Run in CLI: config system session-helper / show / delete 13 / end; config system settings / set sip-expectation disable / set sip-nat-trace disable / end
  5. Keep-Alive Frequency: Set registration expiry and qualify interval to 180 seconds or lower to keep stateful firewall UDP pinholes open.
  6. Firewall Inbound Restrictions: Restrict UDP/TCP port 5060 to carrier registrar IP addresses only. Never leave port 5060 open to 0.0.0.0/0.

Frequently Asked Questions: SIP Trunks in Pakistan

Frequently encountered engineering challenges during telecom SIP trunk commissioning.

Why do calls on PTCL SIP trunks drop at exactly 32 seconds?

A 32-second call drop is the classic symptom of a SIP NAT failure. The call signaling succeeds, but RTP voice audio packets or the SIP ACK response are blocked by your router firewall or corrupted by SIP ALG. Disabling SIP ALG on your ONT/router and forwarding UDP ports 10000–20000 solves the drop.

Which audio codec is mandatory for PTCL and StormFiber SIP trunks?

Pakistani telecom carriers strictly require G.711a (PCMA / A-law) as the first priority codec. Offering G.711u (PCMU) or Opus as primary will result in immediate SIP 488 (Not Acceptable Here) call rejections.

Are my SIP account passwords or credentials stored on your servers?

No. The Arbre SIP Trunk Generator runs 100% locally in your web browser. Credentials never leave your computer, are never sent over the internet, and are not saved in local browser storage.

How do I fix SIP ALG on PTCL Huawei and ZTE GPON routers?

Log into your PTCL ONT admin panel (typically 192.168.1.1), navigate to Security or Application > ALG Configuration, and uncheck SIP ALG. For enterprise deployments, contact PTCL NOC (1218) to put the ONT into Bridge WAN Mode on Port 1 and terminate your public IP on a dedicated firewall.

What Pakistani outbound dial patterns should I configure in Grandstream UCM?

Configure five primary dial patterns: 03XXXXXXXXX (for mobile networks), 02XXXXXXXXX (for NWD landlines), XXXXXXXX with 021 prepend (for 8-digit local Karachi numbers), 00X. (for international calls), and 1X. (for emergency numbers 15, 16, 1122, and 1218).

What is the difference between a Register SIP Trunk and a Peer SIP Trunk in Pakistan?

A Register SIP Trunk authenticates with the carrier using a username and password via periodic SIP REGISTER requests. A Peer SIP Trunk uses direct static IP authentication without registration, commonly used by Nayatel, Cybernet, and corporate PTCL PRI-to-SIP migrations over dedicated fiber VLANs.